From david.osborn at manx.net  Sun Sep  1 16:58:51 2013
From: david.osborn at manx.net (David Osborn)
Date: Sun, 1 Sep 2013 17:58:51 +0100
Subject: [App_rpt-users] Cylon Suppression
In-Reply-To: <522217F2.1090404@gmail.com>
References: <005801cea641$ca22dad0$5e689070$@manx.net>	<CAA_aZLiK0SCQSvz7GYrx4rTdPQQ-XH_QbhgcfsXhkCDVni8WDw@mail.gmail.com>
	<522217F2.1090404@gmail.com>
Message-ID: <004c01cea734$89046e30$9b0d4a90$@manx.net>


"#1) MAJOR SECURITY RED FLAG!!!  You give regular users access to the
OPERATING SYSTEM at the COMMAND LINE LEVEL?"

What? I never said that, and I wouldn't do it. Command line access to the
OS? Do you think I'm that stupid? How would you do that with a DTMF command
(ilink,4 = "Remote Command Mode")? As I understand it, ilink,4 simply relays
DTMF commands from one node to another. I fail to see how that's a security
risk.

 

For info, SSH (command line) access to my nodes has password access disabled
and a pre-shared key is required. Nobody gets a copy of that but me, and
that means that nobody gets "access to the OPERATING SYSTEM at the COMMAND
LINE LEVEL" but me. I don't understand why that got you hyperventilating.
Perhaps you misunderstood.

 

What I'm trying to achieve is design a macro, or macro/script combination
that accepts a DTMF command from a user on one of my local nodes and then
performs the required steps to make an outgoing link from my hub to another
Echolink node.

 

As it happens, I think I've achieved that since I posted. I'm still
fine-tuning, but the user simply enters a short-form DTMF sequence and the
system takes over and sets-up the link. Another short sequence takes the
link down again. All without Cylon activity, and all without anyone ever
getting command-line access.

 

I am fortunate to live in an environment where paranoia is largely
unjustified. In any case, the available commands on my nodes are the usual
basics and practically no-one ever tries them, so they remain as they came
"out of the box" - well, mostly.

 

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From george at dyb.com  Sun Sep  1 17:09:48 2013
From: george at dyb.com (George Csahanin)
Date: Sun, 1 Sep 2013 12:09:48 -0500
Subject: [App_rpt-users] Allstar on different distros
References: <522269A6.4000708@ameritech.net><46AA6F2B-655D-41C1-9EE2-1B3A11E155AB@me.com>
	<52227262.4030801@ameritech.net>
Message-ID: <8E924113B4C1450187B3055687CD9198@lintv.com>

It's all out there. Get the Limey sources and look into the config files and 
Makefiles, eventually it clicks. I have built my own Limey that I needed 
wifi net on and it was a huge learning experience, but it was also fun in a 
kinda sick way.

GeorgeC
W2DB
2360


----- Original Message ----- 
From: "Tony KT9AC" <kt9ac at ameritech.net>
To: "Tim Sawyer" <tim.sawyer at me.com>
Cc: <App_rpt-users at ohnosec.org>
Sent: Saturday, August 31, 2013 5:46 PM
Subject: Re: [App_rpt-users] Allstar on different distros


> Limey only runs on certain motherboards from what is available. Perhaps if 
> it was opened up - some documentation on how to build it for example. I've 
> considered it but can't grasp the cross-compiling methods yet.
>
>
> On 8/31/2013 5:16 PM, Tim Sawyer wrote:
>> My friend in Las Vegas tells me Limey Linux runs on these. He's sending 
>> me one to play with.
>> --
>> Tim
>> :wq
>>
>> On Aug 31, 2013, at 3:09 PM, Tony KT9AC <kt9ac at ameritech.net> wrote:
>>
>>> OK, I'm going to ask if there ongoing efforts to package or provide 
>>> install schemes for different hardware?
>>>
>>> I have had a node running for years on an old P4 just fine. For a 
>>> low-power, fanless box I've gravitated toward the HP Thin clients, which 
>>> really aren't that bad. 1.0Ghz AMD Geode, 512MB ram and an upgrade to a 
>>> 44-pin 8GB Compact Flash. I have a need to build and remotely deploy 
>>> three of these and would like a common install to manage. I do the 
>>> standard noatime and tmpfs changes to keep the flash quiet and happy.
>>>
>>> ACID won't install on Thin Clients; XIPAR will install but takes about 
>>> 3-4 hours and doesn't do MDC1200 or record traffic that I need (acid 
>>> archives). I can install Debian 7.1 in about 20 minutes and it 
>>> absolutely flies. Asterisk 1.8 and the current ACID SVN all compile 
>>> under Debian but I can't get it to work with the URI interface. Dmesg 
>>> and lspci show the URI but Allstar using the "radio tune" shows the card 
>>> as "-1". I have a whole weekend in this latest build attempt.
>>>
>>> It would seem that out in the real world you can install Asterisk on 
>>> anything, and that is the spirit of open-source. What would be required 
>>> to get app_rpt and some of the chan stuff to run on "commodity" installs 
>>> (apt-get install asterisk for example)?
>>>
>>> A lot of this is over my head, but I'm willing to try. I realize that 
>>> some builds were customized for a point in time, so that begs the 
>>> question about modern day attempts (Debian 7.1 runs the 3.2.0-4-686-pae 
>>> kernel). Thanks Kirk for your script, but I found out late last night it 
>>> doesn't use the same hardware that I do.
>>>
>>> Thanks.
>>>
>>> Tony (27129/27418/27460)
>>>
>>>
>>> _______________________________________________
>>> App_rpt-users mailing list
>>> App_rpt-users at ohnosec.org
>>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>>
>
>
> 



From harvard5362 at yahoo.com  Sun Sep  1 20:05:31 2013
From: harvard5362 at yahoo.com (C B)
Date: Sun, 1 Sep 2013 13:05:31 -0700 (PDT)
Subject: [App_rpt-users] linktolink syntax in rpt.conf & passing of tuchtones
Message-ID: <1378065931.55087.YahooMailNeo@web124504.mail.ne1.yahoo.com>

Hi
?
I have read different examples of the?"linktolink" statement in??rpt.conf
?
there are 2 different examples one with a space either side of the "=" sign and one with no spaces.
?
1) linktolinl = yes
2) linktolink=yes
?
Which is the correct syntax.
?
Also when connecting a allstar node to another controllers port, will touch tones frome?a station?conning into a allstar nose?pass from the next?allstar node to the controller?
?
Thank you in advance for your help.
?
Chris
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From harvard5362 at yahoo.com  Sun Sep  1 22:11:43 2013
From: harvard5362 at yahoo.com (C B)
Date: Sun, 1 Sep 2013 15:11:43 -0700 (PDT)
Subject: [App_rpt-users] linktolink syntax in rpt.conf & passing of
	tuchtones
In-Reply-To: <D686E8DD-5323-463A-80F7-6C3FCCA49CC0@aol.com>
References: <1378065931.55087.YahooMailNeo@web124504.mail.ne1.yahoo.com>
	<D686E8DD-5323-463A-80F7-6C3FCCA49CC0@aol.com>
Message-ID: <1378073503.58034.YahooMailNeo@web124501.mail.ne1.yahoo.com>

Thank you
?
Chris
 

________________________________
 From: "cwsaums at aol.com" <cwsaums at aol.com>
To: C B <harvard5362 at yahoo.com> 
Sent: Sunday, September 1, 2013 1:50 PM
Subject: Re: [App_rpt-users] linktolink syntax in rpt.conf & passing of tuchtones
  


Here is the configuration example for value pairs in configuration stanzas
This shows the correct answer is - No spaces

http://ohnosec.org/drupal/node/46

Craig?
AC2FE

Sent from my iPad

On Sep 1, 2013, at 4:05 PM, C B <harvard5362 at yahoo.com> wrote:


Hi

I have read different examples of the?"linktolink" statement in??rpt.conf

there are 2 different examples one with a space either side of the "=" sign and one with no spaces.

1) linktolinl = yes
2) linktolink=yes

Which is the correct syntax.

Also when connecting a allstar node to another controllers port, will touch tones frome?a station?conning into a allstar nose?pass from the next?allstar node to the controller?

Thank you in advance for your help.

Chris
_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From w7ry at centurytel.net  Mon Sep  2 00:15:03 2013
From: w7ry at centurytel.net (Jim W7RY)
Date: Sun, 01 Sep 2013 17:15:03 -0700
Subject: [App_rpt-users] Paul, Your Email box is full
Message-ID: <5223D887.5030405@centurytel.net>

Hey Paul... AH6FM Trying to send you a message... Your email box is 
full... So the system says...

Thanks and Aloha!
73
Jim W7RY



From k6kya at hokeynet.net  Mon Sep  2 22:01:32 2013
From: k6kya at hokeynet.net (Steve Passmore)
Date: Mon, 2 Sep 2013 15:01:32 -0700
Subject: [App_rpt-users] RXONDELAY not working?
In-Reply-To: <BLU180-W59C03DA594F56A9F48D304B1360@phx.gbl>
References: <000d01cea41a$0f3f6180$2dbe2480$@rr.com>
	<AD7FF53AF7954CF4BC214855B106A432@lintv.com>
	<1377824269.70113.YahooMailNeo@web163603.mail.gq1.yahoo.com>
	<000c01cea526$ccf56e60$66e04b20$@rr.com>
	<1377837368.26421.YahooMailNeo@web163605.mail.gq1.yahoo.com>
	<BLU180-W59C03DA594F56A9F48D304B1360@phx.gbl>
Message-ID: <CAMkXzgRJqBeqx+F31u=yNuWRYWp=mFGOCksuS_pxn=QygfPTuA@mail.gmail.com>

Jim,  thanks for fixing that.   I've had a node sitting in the corner for
about 6 months because I needed that function.

Steve


On Fri, Aug 30, 2013 at 6:33 PM, Jim Duuuude <telesistant at hotmail.com>wrote:

>
> Well, it was indeed accurate that:
>
> 1) The "rxondelay" parameter in chan_simpleusb did not work or have any
> effect.
> 2) The software *DID* indeed fully and correctly support the "rxondelay"
> feature, with
>      one very specific exception: The parameter was *NEVER* read from the
> config file!!
>
> All I have to say on my part is: "DUH!!"
>
> The new version (one line change) which fixes this will appear publicly in
> SVN at 7:15 PDT this
> evening.
>
> Jim WB6NIL
>
>
> ------------------------------
> Date: Thu, 29 Aug 2013 21:36:08 -0700
> From: cypresstower at yahoo.com
> To: k5tra at austin.rr.com; george at dyb.com; app_rpt-users at ohnosec.org
>
> Subject: Re: [App_rpt-users] RXONDELAY not working?
>
> Rgr Tom, "I am using usbradio.conf".  Should have realized you were
> referring to.  I have a mobile node with limey linux and simpleusb.conf.  I
> owe you one so I'll test it and see what I come up with.
> Thanks
>
>   *From:* tom <k5tra at austin.rr.com>
> *To:* 'Johnny Keeker' <cypresstower at yahoo.com>; 'George Csahanin' <
> george at dyb.com>; app_rpt-users at ohnosec.org
> *Sent:* Thursday, August 29, 2013 10:15 PM
> *Subject:* RE: [App_rpt-users] RXONDELAY not working?
>
>   Johnny, I think you must be using usbradio instead of simpleusb (as we
> are). Both rxondelay and rxsquelchdelay are parameters in chan_usbradio.c.
> however, rxondelay is in chan_simpleusb.c while rxsquelchdelay is not. The
> problem is rxondelay doesn?t seem to work with simpleusb.
> Tom / K5TRA
>
>  *From:* Johnny Keeker [mailto:cypresstower at yahoo.com]
> *Sent:* Thursday, August 29, 2013 7:58 PM
> *To:* George Csahanin; tom; app_rpt-users at ohnosec.org
> *Subject:* Re: [App_rpt-users] RXONDELAY not working?
>
>    Conducting an experiment with rxondelay, I set my node at 1000ms and
> connected to another node on a different server that I'm able to monitor
> its transmitter.  I applied a signal to the receiver with rxondelay.  The
> transmitter on the connected node keys up and as I count to 10, No audio
> was present so I lowered the rxondelay by half, 500ms. Conducting the same
> test, I found that around the 6 count audio came out the transmitter.
> Lowering the rxondelay to 100ms, and got a count of 2 before hearing
> audio.  I determined that rxondelay is working. In my case it stops the
> ping pong effect. I find a 15ms delay is ideal for a gm300 receiver.  It
> would interesting if someone out there could conduct this same test....  I
> also found using rxsquelchdelay=XXms is another useful setting to eliminate
> any squelch tail noise, but it must be used with dsp or flat audio on ACID.
>   *From:* George Csahanin <george at dyb.com>
> *To:* tom <k5tra at austin.rr.com>; app_rpt-users at ohnosec.org
> *Sent:* Wednesday, August 28, 2013 9:22 PM
> *Subject:* Re: [App_rpt-users] RXONDELAY not working?
>
>   I'll chime in, I have it on a simplex node with simpleusb and it
> doesn't work. Made it 10,000 ms and no effect
>
>  GeorgeC
>  W2DB 2360
>
>  ----- Original Message -----
>  *From:* tom <k5tra at austin.rr.com>
>  *To:* app_rpt-users at ohnosec.org
>  *Sent:* Wednesday, August 28, 2013 1:11 PM
>  *Subject:* Re: [App_rpt-users] RXONDELAY not working?
>
>  The chan_simpleusb.c code clearly *has* the rxondelay parameter.
> However, I tested it on a couple of half-duplex nodes and didn?t see any
> effect. A bug?
>  Tom / K5TRA
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
>
>
> _______________________________________________ App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>


-- 
IMPORTANT: This email is intended for the use of the individual
addressee(s) named above and may contain information that is confidential
privileged or unsuitable for overly sensitive persons with low self-esteem,
no sense of humor or irrational religious beliefs. If you are not the
intended recipient, any dissemination, distribution or copying of this
email is not authorized (either explicitly or implicitly) and constitutes
an irritating social fauxpas. No animals were harmed in the transmission of
this email, although the mutt next door is living on borrowed time, let me
tell you.
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From paulschlaterradio at gmail.com  Tue Sep  3 08:14:35 2013
From: paulschlaterradio at gmail.com (Paul Schlacter)
Date: Tue, 3 Sep 2013 01:14:35 -0700
Subject: [App_rpt-users] execute CLI command via macro?
Message-ID: <CAGD7rLWOSct_1=nEDHnZLzzcG=d03J6GW_Vtxmu7PodBwGOa=g@mail.gmail.com>

Hello Group,
Is there a conventional way to execute CLI commands via a custom macro?

 In other words:

Could app_rpt take a received DTMF command to execute a macro that would
execute a predefined CLI command?

73 all!

Paul
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From bdboyle at bdboyle.com  Tue Sep  3 10:01:53 2013
From: bdboyle at bdboyle.com (Bryan D. Boyle)
Date: Tue, 3 Sep 2013 06:01:53 -0400
Subject: [App_rpt-users] execute CLI command via macro?
In-Reply-To: <CAGD7rLWOSct_1=nEDHnZLzzcG=d03J6GW_Vtxmu7PodBwGOa=g@mail.gmail.com>
References: <CAGD7rLWOSct_1=nEDHnZLzzcG=d03J6GW_Vtxmu7PodBwGOa=g@mail.gmail.com>
Message-ID: <ECF65015-82EF-4D27-A526-6EAF1AF034D6@bdboyle.com>

digging through the list archives, yes it's painful. would yield a treasure trove on this topic.

--
Bryan
Sent from my iPhone 5...small
keyboard, big fingers...please
forgive misspellings...



On Sep 3, 2013, at 4:14, Paul Schlacter <paulschlaterradio at gmail.com> wrote:

> Hello Group, 
> Is there a conventional way to execute CLI commands via a custom macro?
> 
>  In other words:
> 
> Could app_rpt take a received DTMF command to execute a macro that would execute a predefined CLI command?
> 
> 73 all!
> 
> Paul 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users


From ars.w5omr at gmail.com  Tue Sep  3 13:16:33 2013
From: ars.w5omr at gmail.com (Geoff Edmonson)
Date: Tue, 03 Sep 2013 08:16:33 -0500
Subject: [App_rpt-users] execute CLI command via macro?
Message-ID: <chy7485jv0h2stiov55muodq.1378213893577@email.android.com>

You mean like
root@[local]> asterisk -rx "rpt fun [node number] *81"
and... You want to do that from a macro?  No.
However, if you want to connect to different nodes, that's entirely what the [macro] stanza in rpt.conf is all about.

Search the archives here or on ohnosec.org or even Timmy's drupal page. 
Google is your friend.


"Bryan D. Boyle" <bdboyle at bdboyle.com> wrote:

>digging through the list archives, yes it's painful. would yield a treasure trove on this topic.
>
>--
>Bryan
>Sent from my iPhone 5...small
>keyboard, big fingers...please
>forgive misspellings...
>
>
>
>On Sep 3, 2013, at 4:14, Paul Schlacter <paulschlaterradio at gmail.com> wrote:
>
>> Hello Group, 
>> Is there a conventional way to execute CLI commands via a custom macro?
>> 
>>  In other words:
>> 
>> Could app_rpt take a received DTMF command to execute a macro that would execute a predefined CLI command?
>> 
>> 73 all!
>> 
>> Paul 
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

From bdboyle at bdboyle.com  Tue Sep  3 14:13:41 2013
From: bdboyle at bdboyle.com (Bryan D. Boyle)
Date: Tue, 3 Sep 2013 10:13:41 -0400
Subject: [App_rpt-users] execute CLI command via macro?
In-Reply-To: <chy7485jv0h2stiov55muodq.1378213893577@email.android.com>
References: <chy7485jv0h2stiov55muodq.1378213893577@email.android.com>
Message-ID: <AB41BDD4-FCC5-4064-8D50-A9DDE07D0983@bdboyle.com>

uh, you can execute a shell script from a DTMF command by specifying the dtmf in the functions stanza for the node in question:

example:
89=cmd,/path/to/script.sh

where script.sh is something like below located in the /path/to/ directory with the proper chmod values set.

#!/bin/bash
/usr/sbin/asterisk -rx "rpt fun node# *somecommand#"

simple, and fully documented.  actual application is left as an exercise to the reader.  

automating it via a schedule?  use the OS crontab.  

--
Bryan
Sent from my iPhone 5...small
keyboard, big fingers...please
forgive misspellings...



On Sep 3, 2013, at 9:16, Geoff Edmonson <ars.w5omr at gmail.com> wrote:

> You mean like
> root@[local]> asterisk -rx "rpt fun [node number] *81"
> and... You want to do that from a macro?  No.
> However, if you want to connect to different nodes, that's entirely what the [macro] stanza in rpt.conf is all about.
> 
> Search the archives here or on ohnosec.org or even Timmy's drupal page. 
> Google is your friend.
> 
> 
> "Bryan D. Boyle" <bdboyle at bdboyle.com> wrote:
> 
>> digging through the list archives, yes it's painful. would yield a treasure trove on this topic.
>> 
>> --
>> Bryan
>> Sent from my iPhone 5...small
>> keyboard, big fingers...please
>> forgive misspellings...
>> 
>> 
>> 
>> On Sep 3, 2013, at 4:14, Paul Schlacter <paulschlaterradio at gmail.com> wrote:
>> 
>>> Hello Group, 
>>> Is there a conventional way to execute CLI commands via a custom macro?
>>> 
>>> In other words:
>>> 
>>> Could app_rpt take a received DTMF command to execute a macro that would execute a predefined CLI command?
>>> 
>>> 73 all!
>>> 
>>> Paul 
>>> _______________________________________________
>>> App_rpt-users mailing list
>>> App_rpt-users at ohnosec.org
>>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users


From hammin75 at gmail.com  Wed Sep  4 03:05:03 2013
From: hammin75 at gmail.com (Bradley Haney)
Date: Tue, 3 Sep 2013 22:05:03 -0500
Subject: [App_rpt-users] USING own SIP for AUTO PATCH
Message-ID: <CAGjOR2rVeXtN0h9Yw2p0JccZ=YEhxvRSUM9VfZgRYKBGjiPxkQ@mail.gmail.com>

Hello all.   Having an issue with ACID trying to use my own SIP..  It
appears when i try to enter  in the correct info for the peer and registry
statement on the sip.conf and do a astres.sh  it bombs out where for what
ever reason for the next 10 or so  when i log  in and do a asterisk -r  and
do help there are no SIP option on the command line..  then after some time
it some how starts up.   I thought something was wrong with my system so i
put the install disc in again and re downloaded all the software and
started over.  Everything works fine until i try to add a peer into the
SIP.conf  I tried the settings for asterisk 1.2, 1.4  and even 1.8  VOIP
povider i use is callcentric.  Any thoughts?  I have been using callcentric
for about 2 years on my regular PBX and no issues.  ON a side note does
anyone have a good inbound extension.conf  part that woudl include incoming
from my own SIP  to be able to have the asterisk answer and let phone
control work on the repeater.  I would easily pay the 5.00 month for the
service, but our members would like to have a local number for our repeater.

Thanks in advance.
Bradley kc9gqr
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From rpt2 at chuck.midlandsnetworking.com  Wed Sep  4 06:45:00 2013
From: rpt2 at chuck.midlandsnetworking.com (Chuck Henderson)
Date: Wed, 4 Sep 2013 01:45:00 -0500
Subject: [App_rpt-users] USING own SIP for AUTO PATCH
In-Reply-To: <CAGjOR2rVeXtN0h9Yw2p0JccZ=YEhxvRSUM9VfZgRYKBGjiPxkQ@mail.gmail.com>
References: <CAGjOR2rVeXtN0h9Yw2p0JccZ=YEhxvRSUM9VfZgRYKBGjiPxkQ@mail.gmail.com>
Message-ID: <CADEYVwi7Qog9sBJBqSwonbJcvJLsdmPv0VgM=KRmN5D87CUFXw@mail.gmail.com>

Please let me know what you learn on this, as I want to do the same thing
here so that we can have a local number for each of the repeaters we have.



On Tue, Sep 3, 2013 at 10:05 PM, Bradley Haney <hammin75 at gmail.com> wrote:

> Hello all.   Having an issue with ACID trying to use my own SIP..  It
> appears when i try to enter  in the correct info for the peer and registry
> statement on the sip.conf and do a astres.sh  it bombs out where for what
> ever reason for the next 10 or so  when i log  in and do a asterisk -r  and
> do help there are no SIP option on the command line..  then after some time
> it some how starts up.   I thought something was wrong with my system so i
> put the install disc in again and re downloaded all the software and
> started over.  Everything works fine until i try to add a peer into the
> SIP.conf  I tried the settings for asterisk 1.2, 1.4  and even 1.8  VOIP
> povider i use is callcentric.  Any thoughts?  I have been using callcentric
> for about 2 years on my regular PBX and no issues.  ON a side note does
> anyone have a good inbound extension.conf  part that woudl include incoming
> from my own SIP  to be able to have the asterisk answer and let phone
> control work on the repeater.  I would easily pay the 5.00 month for the
> service, but our members would like to have a local number for our repeater.
>
> Thanks in advance.
> Bradley kc9gqr
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
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From chris.andrist at gmail.com  Wed Sep  4 17:19:36 2013
From: chris.andrist at gmail.com (Chris Andrist)
Date: Wed, 4 Sep 2013 11:19:36 -0600
Subject: [App_rpt-users] USING own SIP for AUTO PATCH
In-Reply-To: <mailman.1.1378310402.5731.app_rpt-users@ohnosec.org>
References: <mailman.1.1378310402.5731.app_rpt-users@ohnosec.org>
Message-ID: <-7696893741618382365@unknownmsgid>

Bradley,

I actually wrote something up and put it online. You can see it here:
http://www.allstarnode.com/viewtopic.php?f=6&t=108

Regards,

Chris Andrist, KC7WSU


On Sep 4, 2013, at 10:00 AM, "app_rpt-users-request at ohnosec.org" <
app_rpt-users-request at ohnosec.org> wrote:

2. Re: USING own SIP for AUTO PATCH (Chuck Henderson)


----------------------------------------------------------------------

Message: 1
Date: Tue, 3 Sep 2013 22:05:03 -0500
From: Bradley Haney <hammin75 at gmail.com>
To: app_rpt-users at ohnosec.org
Subject: [App_rpt-users] USING own SIP for AUTO PATCH
Message-ID:
   <CAGjOR2rVeXtN0h9Yw2p0JccZ=YEhxvRSUM9VfZgRYKBGjiPxkQ at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hello all.   Having an issue with ACID trying to use my own SIP..  It
appears when i try to enter  in the correct info for the peer and registry
statement on the sip.conf and do a astres.sh  it bombs out where for what
ever reason for the next 10 or so  when i log  in and do a asterisk -r  and
do help there are no SIP option on the command line..  then after some time
it some how starts up.   I thought something was wrong with my system so i
put the install disc in again and re downloaded all the software and
started over.  Everything works fine until i try to add a peer into the
SIP.conf  I tried the settings for asterisk 1.2, 1.4  and even 1.8  VOIP
povider i use is callcentric.  Any thoughts?  I have been using callcentric
for about 2 years on my regular PBX and no issues.  ON a side note does
anyone have a good inbound extension.conf  part that woudl include incoming
from my own SIP  to be able to have the asterisk answer and let phone
control work on the repeater.  I would easily pay the 5.00 month for the
service, but our members would like to have a local number for our repeater.

Thanks in advance.
Bradley kc9gqr
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From torben at klimt-online.com  Wed Sep  4 20:51:17 2013
From: torben at klimt-online.com (torben at klimt-online.com)
Date: Wed, 4 Sep 2013 22:51:17 +0200
Subject: [App_rpt-users] USING own SIP for AUTO PATCH
In-Reply-To: <-7696893741618382365@unknownmsgid>
References: <mailman.1.1378310402.5731.app_rpt-users@ohnosec.org>
	<-7696893741618382365@unknownmsgid>
Message-ID: <CFFC664EDA2EE5479EA9D2D5A71DAABB53A739A16C@ECCR05PUBLIC.exchange.local>

Hi all,
can somebody tell me what i must write into the extension.conf that every call comming from the sipgate account "go" or answerd by the repeater ?
if it is possible that I can only route spezial numbers to the repeater it will be great

tnx and greetings tom dh6mbt

Von: app_rpt-users-bounces at ohnosec.org [mailto:app_rpt-users-bounces at ohnosec.org] Im Auftrag von Chris Andrist
Gesendet: Mittwoch, 4. September 2013 19:20
An: app_rpt-users at ohnosec.org
Betreff: Re: [App_rpt-users] USING own SIP for AUTO PATCH

Bradley,

I actually wrote something up and put it online. You can see it here: http://www.allstarnode.com/viewtopic.php?f=6&t=108

Regards,

Chris Andrist, KC7WSU


On Sep 4, 2013, at 10:00 AM, "app_rpt-users-request at ohnosec.org<mailto:app_rpt-users-request at ohnosec.org>" <app_rpt-users-request at ohnosec.org<mailto:app_rpt-users-request at ohnosec.org>> wrote:
2. Re: USING own SIP for AUTO PATCH (Chuck Henderson)


----------------------------------------------------------------------

Message: 1
Date: Tue, 3 Sep 2013 22:05:03 -0500
From: Bradley Haney <hammin75 at gmail.com<mailto:hammin75 at gmail.com>>
To: app_rpt-users at ohnosec.org<mailto:app_rpt-users at ohnosec.org>
Subject: [App_rpt-users] USING own SIP for AUTO PATCH
Message-ID:
   <CAGjOR2rVeXtN0h9Yw2p0JccZ=YEhxvRSUM9VfZgRYKBGjiPxkQ at mail.gmail.com<mailto:CAGjOR2rVeXtN0h9Yw2p0JccZ=YEhxvRSUM9VfZgRYKBGjiPxkQ at mail.gmail.com>>
Content-Type: text/plain; charset="iso-8859-1"

Hello all.   Having an issue with ACID trying to use my own SIP..  It
appears when i try to enter  in the correct info for the peer and registry
statement on the sip.conf and do a astres.sh  it bombs out where for what
ever reason for the next 10 or so  when i log  in and do a asterisk -r  and
do help there are no SIP option on the command line..  then after some time
it some how starts up.   I thought something was wrong with my system so i
put the install disc in again and re downloaded all the software and
started over.  Everything works fine until i try to add a peer into the
SIP.conf  I tried the settings for asterisk 1.2, 1.4  and even 1.8  VOIP
povider i use is callcentric.  Any thoughts?  I have been using callcentric
for about 2 years on my regular PBX and no issues.  ON a side note does
anyone have a good inbound extension.conf  part that woudl include incoming
from my own SIP  to be able to have the asterisk answer and let phone
control work on the repeater.  I would easily pay the 5.00 month for the
service, but our members would like to have a local number for our repeater.

Thanks in advance.
Bradley kc9gqr
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From ve3elb at yahoo.com  Thu Sep  5 03:10:15 2013
From: ve3elb at yahoo.com (Vince P)
Date: Wed, 4 Sep 2013 23:10:15 -0400
Subject: [App_rpt-users] Setting System Time
Message-ID: <CD88227D-950D-4233-AD99-673A1670AC31@yahoo.com>

Hello all. Can anyone guide me on to how to setting up the time on asterisk. The time was set but for some reason a few weeks ago I noticed that its off by 8 hours. Thank you all for your help. 

73, Vince VE3ELB

From cypresstower at yahoo.com  Thu Sep  5 04:18:26 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Wed, 4 Sep 2013 21:18:26 -0700 (PDT)
Subject: [App_rpt-users] Setting System Time
In-Reply-To: <CD88227D-950D-4233-AD99-673A1670AC31@yahoo.com>
References: <CD88227D-950D-4233-AD99-673A1670AC31@yahoo.com>
Message-ID: <1378354706.75967.YahooMailNeo@web163602.mail.gq1.yahoo.com>

Set up Timezone Carfully Copy each line below to your linux box as root.? 
Before begining, check /usr/share/zoneinfo/ for your time zone. 
Enter your time zone, a?space, followed by the word localtime
The example below is if you lived in Toronto Canada.
?
[root at node /]#cd /etc 
[root at node /]#cp localtime localtime.orig 
[root at node /]#ln ?-sf /usr/share/zoneinfo/Canada/Eastern localtime 
?
Check your results 
[root at node /]#date 
?
Synchronize time from the web
[root at node /]#ntpdate pool.ntp.org 
?
Set Server to Auto Sync The Time from the web 
[root at node /]#ntpdate 0.us.pool.ntp.org 
[root at node /]#hwclock --systohc 
?
Restart ntpd service
[root at node /]#service ntpd start 
?
Enable service at startup
[root at node /]#chkconfig ntpd on

________________________________
From: Vince P <ve3elb at yahoo.com>
To: APP RPT Forums <app_rpt-users at ohnosec.org> 
Sent: Wednesday, September 4, 2013 11:10 PM
Subject: [App_rpt-users] Setting System Time


Hello all. Can anyone guide me on to how to setting up the time on asterisk. The time was set but for some reason a few weeks ago I noticed that its off by 8 hours. Thank you all for your help. 

73, Vince VE3ELB
_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From george at dyb.com  Thu Sep  5 05:55:39 2013
From: george at dyb.com (George Csahanin)
Date: Thu, 5 Sep 2013 00:55:39 -0500
Subject: [App_rpt-users] Setting System Time
References: <CD88227D-950D-4233-AD99-673A1670AC31@yahoo.com>
	<1378354706.75967.YahooMailNeo@web163602.mail.gq1.yahoo.com>
Message-ID: <9354E64275F54A2A880370A511CEA2CB@lintv.com>

Except Limey. The timezone info isn't there. I just live with it. I was 
using a timesserver and if I set the system time with a fudge factor to be 
right as soon as the box sync-ed to network time it was off again.

But in ACID, yup, you have it right

GeorgeC
W2DB
2360

----- Original Message ----- 
From: Johnny Keeker
To: Vince P ; APP RPT Forums
Sent: Wednesday, September 04, 2013 11:18 PM
Subject: Re: [App_rpt-users] Setting System Time


Set up Timezone Carfully Copy each line below to your linux box as root.
Before begining, check /usr/share/zoneinfo/ for your time zone.
Enter your time zone, a space, followed by the word localtime
The example below is if you lived in Toronto Canada.

[root at node /]#cd /etc
[root at node /]#cp localtime localtime.orig
[root at node /]#ln ?-sf /usr/share/zoneinfo/Canada/Eastern localtime

Check your results
[root at node /]#date

Synchronize time from the web
[root at node /]#ntpdate pool.ntp.org

Set Server to Auto Sync The Time from the web
[root at node /]#ntpdate 0.us.pool.ntp.org
[root at node /]#hwclock --systohc

Restart ntpd service
[root at node /]#service ntpd start

Enable service at startup
[root at node /]#chkconfig ntpd on


From: Vince P <ve3elb at yahoo.com>
To: APP RPT Forums <app_rpt-users at ohnosec.org>
Sent: Wednesday, September 4, 2013 11:10 PM
Subject: [App_rpt-users] Setting System Time


Hello all. Can anyone guide me on to how to setting up the time on asterisk. 
The time was set but for some reason a few weeks ago I noticed that its off 
by 8 hours. Thank you all for your help.

73, Vince VE3ELB
_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 



From rpt2 at chuck.midlandsnetworking.com  Thu Sep  5 06:03:06 2013
From: rpt2 at chuck.midlandsnetworking.com (Chuck Henderson)
Date: Thu, 5 Sep 2013 01:03:06 -0500
Subject: [App_rpt-users] Setting System Time
In-Reply-To: <9354E64275F54A2A880370A511CEA2CB@lintv.com>
References: <CD88227D-950D-4233-AD99-673A1670AC31@yahoo.com>
	<1378354706.75967.YahooMailNeo@web163602.mail.gq1.yahoo.com>
	<9354E64275F54A2A880370A511CEA2CB@lintv.com>
Message-ID: <CADEYVwh9CRqVfktC+deOxPyNwHJtSTnR5X7mo7+jdTsqu0zAHQ@mail.gmail.com>

On anything but the fastest multicore systems , I have found that when
using usbradio and running ntpd, it will cause anomalies in the repeater
sound.  My solution is to not run ntpd, but to just have a cron job that
runs ntpdate once an hour.  That way there is only one glitch in the sound
each hour.  Your mileage may vary.


On Thu, Sep 5, 2013 at 12:55 AM, George Csahanin <george at dyb.com> wrote:

> Except Limey. The timezone info isn't there. I just live with it. I was
> using a timesserver and if I set the system time with a fudge factor to be
> right as soon as the box sync-ed to network time it was off again.
>
> But in ACID, yup, you have it right
>
> GeorgeC
> W2DB
> 2360
>
> ----- Original Message ----- From: Johnny Keeker
> To: Vince P ; APP RPT Forums
> Sent: Wednesday, September 04, 2013 11:18 PM
> Subject: Re: [App_rpt-users] Setting System Time
>
>
>
> Set up Timezone Carfully Copy each line below to your linux box as root.
> Before begining, check /usr/share/zoneinfo/ for your time zone.
> Enter your time zone, a space, followed by the word localtime
> The example below is if you lived in Toronto Canada.
>
> [root at node /]#cd /etc
> [root at node /]#cp localtime localtime.orig
> [root at node /]#ln ?-sf /usr/share/zoneinfo/Canada/**Eastern localtime
>
> Check your results
> [root at node /]#date
>
> Synchronize time from the web
> [root at node /]#ntpdate pool.ntp.org
>
> Set Server to Auto Sync The Time from the web
> [root at node /]#ntpdate 0.us.pool.ntp.org
> [root at node /]#hwclock --systohc
>
> Restart ntpd service
> [root at node /]#service ntpd start
>
> Enable service at startup
> [root at node /]#chkconfig ntpd on
>
>
> From: Vince P <ve3elb at yahoo.com>
> To: APP RPT Forums <app_rpt-users at ohnosec.org>
> Sent: Wednesday, September 4, 2013 11:10 PM
> Subject: [App_rpt-users] Setting System Time
>
>
> Hello all. Can anyone guide me on to how to setting up the time on
> asterisk. The time was set but for some reason a few weeks ago I noticed
> that its off by 8 hours. Thank you all for your help.
>
> 73, Vince VE3ELB
> ______________________________**_________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/**mailman/listinfo/app_rpt-users<http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users>
> ______________________________**_________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/**mailman/listinfo/app_rpt-users<http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users>
>
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From george at dyb.com  Thu Sep  5 06:09:35 2013
From: george at dyb.com (George Csahanin)
Date: Thu, 5 Sep 2013 01:09:35 -0500
Subject: [App_rpt-users] Setting System Time
References: <CD88227D-950D-4233-AD99-673A1670AC31@yahoo.com>
	<1378354706.75967.YahooMailNeo@web163602.mail.gq1.yahoo.com>
	<9354E64275F54A2A880370A511CEA2CB@lintv.com>
	<CADEYVwh9CRqVfktC+deOxPyNwHJtSTnR5X7mo7+jdTsqu0zAHQ@mail.gmail.com>
Message-ID: <C0D32F5D758C4CCBA5E18D32159EAFEE@lintv.com>

NOW THAT's interesting. That might have been my usbradio proble,. Converted to simpleusb and use the summed COS+CG decode in the GE MVP I have it on now, so not an issue...but...I need to experiment with that.

GeorgeC
  ----- Original Message ----- 
  From: Chuck Henderson 
  To: George Csahanin 
  Cc: Johnny Keeker ; Vince P ; APP RPT Forums 
  Sent: Thursday, September 05, 2013 1:03 AM
  Subject: Re: [App_rpt-users] Setting System Time


  On anything but the fastest multicore systems , I have found that when using usbradio and running ntpd, it will cause anomalies in the repeater sound.  My solution is to not run ntpd, but to just have a cron job that runs ntpdate once an hour.  That way there is only one glitch in the sound each hour.  Your mileage may vary.




  On Thu, Sep 5, 2013 at 12:55 AM, George Csahanin <george at dyb.com> wrote:

    Except Limey. The timezone info isn't there. I just live with it. I was using a timesserver and if I set the system time with a fudge factor to be right as soon as the box sync-ed to network time it was off again.

    But in ACID, yup, you have it right

    GeorgeC
    W2DB
    2360

    ----- Original Message ----- From: Johnny Keeker
    To: Vince P ; APP RPT Forums
    Sent: Wednesday, September 04, 2013 11:18 PM
    Subject: Re: [App_rpt-users] Setting System Time



    Set up Timezone Carfully Copy each line below to your linux box as root.
    Before begining, check /usr/share/zoneinfo/ for your time zone.
    Enter your time zone, a space, followed by the word localtime
    The example below is if you lived in Toronto Canada.

    [root at node /]#cd /etc
    [root at node /]#cp localtime localtime.orig
    [root at node /]#ln ?-sf /usr/share/zoneinfo/Canada/Eastern localtime

    Check your results
    [root at node /]#date

    Synchronize time from the web
    [root at node /]#ntpdate pool.ntp.org

    Set Server to Auto Sync The Time from the web
    [root at node /]#ntpdate 0.us.pool.ntp.org
    [root at node /]#hwclock --systohc

    Restart ntpd service
    [root at node /]#service ntpd start

    Enable service at startup
    [root at node /]#chkconfig ntpd on


    From: Vince P <ve3elb at yahoo.com>
    To: APP RPT Forums <app_rpt-users at ohnosec.org>
    Sent: Wednesday, September 4, 2013 11:10 PM
    Subject: [App_rpt-users] Setting System Time


    Hello all. Can anyone guide me on to how to setting up the time on asterisk. The time was set but for some reason a few weeks ago I noticed that its off by 8 hours. Thank you all for your help.

    73, Vince VE3ELB
    _______________________________________________
    App_rpt-users mailing list
    App_rpt-users at ohnosec.org
    http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 
    _______________________________________________
    App_rpt-users mailing list
    App_rpt-users at ohnosec.org
    http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users


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From ken-robinson at o2.co.uk  Thu Sep  5 08:32:05 2013
From: ken-robinson at o2.co.uk (Ken Robinson)
Date: Thu, 5 Sep 2013 09:32:05 +0100
Subject: [App_rpt-users] Beagle board Black
Message-ID: <2e4001ceaa12$677ea790$367bf6b0$@o2.co.uk>

 

Hi All,

 

Can anyone tell me if the Beagle board Black will act as an Allstar node,
reading about it I think some of the input outputs have been 

taken out of the new board

 

73's 

Ken 

G0lce

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From wbs099 at yahoo.com  Thu Sep  5 12:18:58 2013
From: wbs099 at yahoo.com (Bill South)
Date: Thu, 5 Sep 2013 05:18:58 -0700 (PDT)
Subject: [App_rpt-users] SIP VoIP for Asterisk
Message-ID: <1378383538.16999.YahooMailBasic@web161403.mail.bf1.yahoo.com>

     I'm thinking of adding some type of SIP trunking or other VoIP service provider to my ACID Asterisk system to support in/out bound calling.  I've read some emails on the app_rpt reflector with names of providers mentioned, but I am looking for recommendations, as there are zillions of VoIP providers out there.  This is going to be used very sparingly so least-cost is a good thing, but good reliability and no bombardment with email adds by the provider is desired too.  I can easily get by with a single number, but may want to add additional DIDs later.  Thoughts?

Bill




From tim.sawyer at me.com  Thu Sep  5 14:17:02 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Thu, 05 Sep 2013 07:17:02 -0700
Subject: [App_rpt-users] SIP VoIP for Asterisk
In-Reply-To: <1378383538.16999.YahooMailBasic@web161403.mail.bf1.yahoo.com>
References: <1378383538.16999.YahooMailBasic@web161403.mail.bf1.yahoo.com>
Message-ID: <E25CD727-9DD8-4BCE-9A6B-57C19E1053E4@me.com>

I've been using Vitelity for a few years and am happy with them. They charge $1.49 per month per DID and 1.2 cents per minute. 

--
Tim
:wq

On Sep 5, 2013, at 5:18 AM, Bill South <wbs099 at yahoo.com> wrote:

>     I'm thinking of adding some type of SIP trunking or other VoIP service provider to my ACID Asterisk system to support in/out bound calling.  I've read some emails on the app_rpt reflector with names of providers mentioned, but I am looking for recommendations, as there are zillions of VoIP providers out there.  This is going to be used very sparingly so least-cost is a good thing, but good reliability and no bombardment with email adds by the provider is desired too.  I can easily get by with a single number, but may want to add additional DIDs later.  Thoughts?
> 
> Bill
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



From DwaineGarden at rogers.com  Thu Sep  5 16:13:41 2013
From: DwaineGarden at rogers.com (Dwaine Garden VE3GIF)
Date: Thu, 05 Sep 2013 12:13:41 -0400
Subject: [App_rpt-users] SIP VoIP for Asterisk
Message-ID: <61soqbviophkjxciig46chmc.1378397621879@email.android.com>

It works great until the hacks find the machine.  They port scan non stop.  Its especially fun when their scripts dial 911 constantly.  There is no way to turn off dialing 911 for SIP.

Bill South <wbs099 at yahoo.com> wrote:

>     I'm thinking of adding some type of SIP trunking or other VoIP service provider to my ACID Asterisk system to support in/out bound calling.  I've read some emails on the app_rpt reflector with names of providers mentioned, but I am looking for recommendations, as there are zillions of VoIP providers out there.  This is going to be used very sparingly so least-cost is a good thing, but good reliability and no bombardment with email adds by the provider is desired too.  I can easily get by with a single number, but may want to add additional DIDs later.  Thoughts?
>
>Bill
>
>
>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

From dshaw at ke6upi.com  Thu Sep  5 16:58:56 2013
From: dshaw at ke6upi.com (David KE6UPI)
Date: Thu, 5 Sep 2013 09:58:56 -0700
Subject: [App_rpt-users] SIP VoIP for Asterisk
In-Reply-To: <61soqbviophkjxciig46chmc.1378397621879@email.android.com>
References: <61soqbviophkjxciig46chmc.1378397621879@email.android.com>
Message-ID: <CAH4RvbLT5vz90AM2vHKp_omF8gn0X_spuWMXnPsqKuMFA+koZw@mail.gmail.com>

I'm sorry Dwaine what are you talking about? Sorry If I don't understand
what your talking about.

I have both a public Asterisk server and a local Asterisk server. I have
never had anyone connect and make a call that was not authenticated user..
Yes they try and fail2ban will block them. There are many way to stop
unwanted hackers on your server.


As for dialing 911 just make a dial plain to route to space if you want.

Google "Asterisk Security"

http://www.voip-info.org/wiki/view/Asterisk+security


David



On Thu, Sep 5, 2013 at 9:13 AM, Dwaine Garden VE3GIF <
DwaineGarden at rogers.com> wrote:

> It works great until the hacks find the machine.  They port scan non stop.
>  Its especially fun when their scripts dial 911 constantly.  There is no
> way to turn off dialing 911 for SIP.
>
> Bill South <wbs099 at yahoo.com> wrote:
>
> >     I'm thinking of adding some type of SIP trunking or other VoIP
> service provider to my ACID Asterisk system to support in/out bound
> calling.  I've read some emails on the app_rpt reflector with names of
> providers mentioned, but I am looking for recommendations, as there are
> zillions of VoIP providers out there.  This is going to be used very
> sparingly so least-cost is a good thing, but good reliability and no
> bombardment with email adds by the provider is desired too.  I can easily
> get by with a single number, but may want to add additional DIDs later.
>  Thoughts?
> >
> >Bill
> >
> >
> >_______________________________________________
> >App_rpt-users mailing list
> >App_rpt-users at ohnosec.org
> >http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
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From n5zua at earthlink.net  Thu Sep  5 16:59:44 2013
From: n5zua at earthlink.net (Steve Agee)
Date: Thu, 5 Sep 2013 11:59:44 -0500
Subject: [App_rpt-users] SIP VoIP for Asterisk
References: <61soqbviophkjxciig46chmc.1378397621879@email.android.com>
Message-ID: <6989AB4F31BA4BF9AB490D550BA5E261@steveea3dc3d27>

I change my SIP port from the default 5060 to 15060 and have not had any 
issues since. It seems that (most of the time) they only try the default.

N5ZUA

----- Original Message ----- 
From: "Dwaine Garden VE3GIF" <DwaineGarden at rogers.com>
To: "Bill South" <wbs099 at yahoo.com>
Cc: <app_rpt-users at ohnosec.org>
Sent: Thursday, September 05, 2013 11:13 AM
Subject: Re: [App_rpt-users] SIP VoIP for Asterisk


> It works great until the hacks find the machine.  They port scan non stop. 
> Its especially fun when their scripts dial 911 constantly.  There is no 
> way to turn off dialing 911 for SIP.
>
> Bill South <wbs099 at yahoo.com> wrote:
>
>>     I'm thinking of adding some type of SIP trunking or other VoIP 
>> service provider to my ACID Asterisk system to support in/out bound 
>> calling.  I've read some emails on the app_rpt reflector with names of 
>> providers mentioned, but I am looking for recommendations, as there are 
>> zillions of VoIP providers out there.  This is going to be used very 
>> sparingly so least-cost is a good thing, but good reliability and no 
>> bombardment with email adds by the provider is desired too.  I can easily 
>> get by with a single number, but may want to add additional DIDs later. 
>> Thoughts?
>>
>>Bill
>>
>>
>>_______________________________________________
>>App_rpt-users mailing list
>>App_rpt-users at ohnosec.org
>>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 



From DwaineGarden at rogers.com  Thu Sep  5 18:09:59 2013
From: DwaineGarden at rogers.com (Dwaine Garden VE3GIF)
Date: Thu, 05 Sep 2013 14:09:59 -0400
Subject: [App_rpt-users] SIP VoIP for Asterisk
Message-ID: <xn0rojqy5gpqhtreyrdsbhyg.1378404599863@email.android.com>

I changed the port.   They just kept scanning every IP port for the IP address.   I had to ask Rogers for another dhcp IP address.

The whole time they never got into the box.  LOL....

Steve Agee <n5zua at earthlink.net> wrote:

>I change my SIP port from the default 5060 to 15060 and have not had any 
>issues since. It seems that (most of the time) they only try the default.
>
>N5ZUA
>
>----- Original Message ----- 
>From: "Dwaine Garden VE3GIF" <DwaineGarden at rogers.com>
>To: "Bill South" <wbs099 at yahoo.com>
>Cc: <app_rpt-users at ohnosec.org>
>Sent: Thursday, September 05, 2013 11:13 AM
>Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
>
>
>> It works great until the hacks find the machine.  They port scan non stop. 
>> Its especially fun when their scripts dial 911 constantly.  There is no 
>> way to turn off dialing 911 for SIP.
>>
>> Bill South <wbs099 at yahoo.com> wrote:
>>
>>>     I'm thinking of adding some type of SIP trunking or other VoIP 
>>> service provider to my ACID Asterisk system to support in/out bound 
>>> calling.  I've read some emails on the app_rpt reflector with names of 
>>> providers mentioned, but I am looking for recommendations, as there are 
>>> zillions of VoIP providers out there.  This is going to be used very 
>>> sparingly so least-cost is a good thing, but good reliability and no 
>>> bombardment with email adds by the provider is desired too.  I can easily 
>>> get by with a single number, but may want to add additional DIDs later. 
>>> Thoughts?
>>>
>>>Bill
>>>
>>>
>>>_______________________________________________
>>>App_rpt-users mailing list
>>>App_rpt-users at ohnosec.org
>>>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 
>

From DwaineGarden at rogers.com  Thu Sep  5 18:07:58 2013
From: DwaineGarden at rogers.com (Dwaine Garden VE3GIF)
Date: Thu, 05 Sep 2013 14:07:58 -0400
Subject: [App_rpt-users] SIP VoIP for Asterisk
Message-ID: <f23kgbt8i5o96yxhab0lkale.1378404056404@email.android.com>

The problem is you are not allowed by law to have a phone without unrestricted access to 911.  I had Metro Toronto police at my door explaining that even if I block 911 to any outside connections I would be breaking the law.  If you have a server on the internet with sip.  They have to able to connect to be able to call 911. 

I told the police it was retard.  They told me that was fine they will charge me.

Police told me that even if someone breaks into your house.  If there is a phone install,  the criminals better have access to dial 911 unrestrictive.

The hackers did not get into the box.  They were trying for months.   Got pissed off and changed their script to dial 911 constantly.  SIP and DID see a 911 call.  It dials it.  No questions asked.  No login or nothing.

The Police told me it was a huge problem.  SIP or DID are setup like a public pay phone.  Full access to 911.

It was an eye opener for me.   You learn something new everyday.  If I see someone asking about SIP or DID.  I let them know about my experience.

David KE6UPI <dshaw at ke6upi.com> wrote:

>I'm sorry Dwaine what are you talking about? Sorry If I don't understand what your talking about.
>
>I have both a public Asterisk server and a local Asterisk server. I have never had anyone connect and make a call that was not authenticated user.. Yes they try and fail2ban will block them. There are many way to stop unwanted hackers on your server. 
>
>
>As for dialing 911 just make a dial plain to route to space if you want. 
>
>Google "Asterisk Security"
>
>
>http://www.voip-info.org/wiki/view/Asterisk+security
>
>
>David
>
>
>
>
>On Thu, Sep 5, 2013 at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:
>
>It works great until the hacks find the machine. ?They port scan non stop. ?Its especially fun when their scripts dial 911 constantly. ?There is no way to turn off dialing 911 for SIP.
>
>
>Bill South <wbs099 at yahoo.com> wrote:
>
>> ? ? I'm thinking of adding some type of SIP trunking or other VoIP service provider to my ACID Asterisk system to support in/out bound calling. ?I've read some emails on the app_rpt reflector with names of providers mentioned, but I am looking for recommendations, as there are zillions of VoIP providers out there. ?This is going to be used very sparingly so least-cost is a good thing, but good reliability and no bombardment with email adds by the provider is desired too. ?I can easily get by with a single number, but may want to add additional DIDs later. ?Thoughts?
>>
>>Bill
>>
>>
>>_______________________________________________
>>App_rpt-users mailing list
>>App_rpt-users at ohnosec.org
>>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
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From N7LD at aol.com  Thu Sep  5 18:25:27 2013
From: N7LD at aol.com (N7LD at aol.com)
Date: Thu, 5 Sep 2013 14:25:27 -0400 (EDT)
Subject: [App_rpt-users] permit=  echolink.conf
Message-ID: <bf04a.50e55d88.3f5a2697@aol.com>

Looks like I am having the same problem. Looks like the MAX stations is  30
Lee
 
 
 
In a message dated 5/21/2013 11:08:47 A.M. Pacific Daylight Time,  
w8jtw at yahoo.com writes:

Is there  anyway to increase the permit=  list in echolink.conf to work 
with more  than 25 or so stations?  Once I get beyond 25 it does not want to 
allow  beyond the first 25.  Or is there a way to limit to just USA 
connections?  
I dont want to run a wide open echolink but have many  friends i don't want 
to block    need to have capacity of about 100  or so.

Thought of trying Permit=W*,N*,A*,K* not sure if that would  limit to just 
US callsigns or not.

Any  suggestions?

Thanks

Joe 
W8JTW
Node 27891   
_______________________________________________
App_rpt-users mailing  list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From telesistant at hotmail.com  Thu Sep  5 18:29:44 2013
From: telesistant at hotmail.com (Jim Duuuude)
Date: Thu, 5 Sep 2013 11:29:44 -0700
Subject: [App_rpt-users] SIP VoIP for Asterisk
In-Reply-To: <f23kgbt8i5o96yxhab0lkale.1378404056404@email.android.com>
References: <f23kgbt8i5o96yxhab0lkale.1378404056404@email.android.com>
Message-ID: <BLU180-W4989AF78DEA16E83A66449B1330@phx.gbl>

Okay (and yes, that is STUPID and most likely WRONG, but most police depts are
just completely ignorant of telecom issues). So, fine... GIVE them access to 911.
Let them dial it. But, sadly, on YOUR phone network, the dialing string is just a LITTLE
bit longer (like about 30 digits in front of the 911)... get it?

Jim


Date: Thu, 5 Sep 2013 14:07:58 -0400
From: DwaineGarden at rogers.com
To: dshaw at ke6upi.com
CC: app_rpt-users at ohnosec.org
Subject: Re: [App_rpt-users] SIP VoIP for Asterisk

The problem is you are not allowed by law to have a phone without unrestricted access to 911.  I had Metro Toronto police at my door explaining that even if I block 911 to any outside connections I would be breaking the law.  If you have a server on the internet with sip.  They have to able to connect to be able to call 911. 

I told the police it was retard.  They told me that was fine they will charge me.

Police told me that even if someone breaks into your house.  If there is a phone install,  the criminals better have access to dial 911 unrestrictive.

The hackers did not get into the box.  They were trying for months.   Got pissed off and changed their script to dial 911 constantly.  SIP and DID see a 911 call.  It dials it.  No questions asked.  No login or nothing.

The Police told me it was a huge problem.  SIP or DID are setup like a public pay phone.  Full access to 911.

It was an eye opener for me.   You learn something new everyday.  If I see someone asking about SIP or DID.  I let them know about my experience.

David KE6UPI <dshaw at ke6upi.com> wrote:

I'm sorry Dwaine what are you talking about? Sorry If I don't understand what your talking about.

I have both a public Asterisk server and a local Asterisk server. I have never had anyone connect and make a call that was not authenticated user.. Yes they try and fail2ban will block them. There are many way to stop unwanted hackers on your server. 



As for dialing 911 just make a dial plain to route to space if you want. 

Google "Asterisk Security"

http://www.voip-info.org/wiki/view/Asterisk+security



David



On Thu, Sep 5, 2013 at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:

It works great until the hacks find the machine.  They port scan non stop.  Its especially fun when their scripts dial 911 constantly.  There is no way to turn off dialing 911 for SIP.




Bill South <wbs099 at yahoo.com> wrote:



>     I'm thinking of adding some type of SIP trunking or other VoIP service provider to my ACID Asterisk system to support in/out bound calling.  I've read some emails on the app_rpt reflector with names of providers mentioned, but I am looking for recommendations, as there are zillions of VoIP providers out there.  This is going to be used very sparingly so least-cost is a good thing, but good reliability and no bombardment with email adds by the provider is desired too.  I can easily get by with a single number, but may want to add additional DIDs later.  Thoughts?


>

>Bill

>

>

>_______________________________________________

>App_rpt-users mailing list

>App_rpt-users at ohnosec.org

>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

_______________________________________________

App_rpt-users mailing list

App_rpt-users at ohnosec.org

http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users




_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 		 	   		  
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From DwaineGarden at rogers.com  Thu Sep  5 18:58:42 2013
From: DwaineGarden at rogers.com (Dwaine Garden VE3GIF)
Date: Thu, 05 Sep 2013 14:58:42 -0400
Subject: [App_rpt-users] SIP VoIP for Asterisk
Message-ID: <p4w6jrfo77byin76d5emwfps.1378407131503@email.android.com>

It was an interesting experience.  Has anyone else experienced the same problem and had the police involved?  
I did mention that I would change the
Dial string.  The police told me that a person has to be able to dial 911 and get emergency services.

I gave up and just walked over to the all-star computer and hit the power button.  Sadly, it has been off ever since.

If I had changed the 911 Dial string before the incident.  The police would of not gotten involved. Never known about the situation.

Just let everyone know.  The two police officers were very good about it.  They told me that its happening a lot and 911 is getting a little upset about it.

I told them they should go after the person responsible.  They told me that's you.   The line is in your name.

Anyway, just wanted to throw out there my experience.

Dwaine

Jim Duuuude <telesistant at hotmail.com> wrote:

><!-- .hmmessage P { margin:0px; padding:0px } body.hmmessage { font-size: 12pt; font-family:Calibri } -->
>
>Okay (and yes, that is STUPID and most likely WRONG, but most police depts are
>just completely ignorant of telecom issues). So, fine... GIVE them access to 911.
>Let them dial it. But, sadly, on YOUR phone network, the dialing string is just a LITTLE
>bit longer (like about 30 digits in front of the 911)... get it?
>
>Jim
>
>
>Date: Thu, 5 Sep 2013 14:07:58 -0400
>From: DwaineGarden at rogers.com
>To: dshaw at ke6upi.com
>CC: app_rpt-users at ohnosec.org
>Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
>
>The problem is you are not allowed by law to have a phone without unrestricted access to 911. I had Metro Toronto police at my door explaining that even if I block 911 to any outside connections I would be breaking the law. If you have a server on the internet with sip. They have to able to connect to be able to call 911. 
>
>I told the police it was retard. They told me that was fine they will charge me.
>
>Police told me that even if someone breaks into your house. If there is a phone install, the criminals better have access to dial 911 unrestrictive.
>
>The hackers did not get into the box. They were trying for months. Got pissed off and changed their script to dial 911 constantly. SIP and DID see a 911 call. It dials it. No questions asked. No login or nothing.
>
>The Police told me it was a huge problem. SIP or DID are setup like a public pay phone. Full access to 911.
>
>It was an eye opener for me. You learn something new everyday. If I see someone asking about SIP or DID. I let them know about my experience.
>
>David KE6UPI <dshaw at ke6upi.com> wrote:
>
>I'm sorry Dwaine what are you talking about? Sorry If I don't understand what your talking about.
>
>I have both a public Asterisk server and a local Asterisk server. I have never had anyone connect and make a call that was not authenticated user.. Yes they try and fail2ban will block them. There are many way to stop unwanted hackers on your server. 
>
>
>As for dialing 911 just make a dial plain to route to space if you want. 
>
>Google "Asterisk Security"
>
>
>http://www.voip-info.org/wiki/view/Asterisk+security
>
>
>David
>
>
>
>
>On Thu, Sep 5, 2013 at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:
>
>It works great until the hacks find the machine. ?They port scan non stop. ?Its especially fun when their scripts dial 911 constantly. ?There is no way to turn off dialing 911 for SIP.
>
>
>Bill South <wbs099 at yahoo.com> wrote:
>
>> ? ? I'm thinking of adding some type of SIP trunking or other VoIP service provider to my ACID Asterisk system to support in/out bound calling. ?I've read some emails on the app_rpt reflector with names of providers mentioned, but I am looking for recommendations, as there are zillions of VoIP providers out there. ?This is going to be used very sparingly so least-cost is a good thing, but good reliability and no bombardment with email adds by the provider is desired too. ?I can easily get by with a single number, but may want to add additional DIDs later. ?Thoughts?
>>
>>Bill
>>
>>
>>_______________________________________________
>>App_rpt-users mailing list
>>App_rpt-users at ohnosec.org
>>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
>
>_______________________________________________ App_rpt-users mailing list App_rpt-users at ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>		 	 		 
>
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From telesistant at hotmail.com  Thu Sep  5 19:24:28 2013
From: telesistant at hotmail.com (Jim Duuuude)
Date: Thu, 5 Sep 2013 12:24:28 -0700
Subject: [App_rpt-users] SIP VoIP for Asterisk
In-Reply-To: <p4w6jrfo77byin76d5emwfps.1378407131503@email.android.com>
References: <p4w6jrfo77byin76d5emwfps.1378407131503@email.android.com>
Message-ID: <BLU180-W754E1912C04A86119FB648B1330@phx.gbl>

Double BRAVO-SIERRA!!

If that were true, a LOT of large business owners (some of which may even operate
large businesses :-) ) would be in jail, if they HAPPEN to have a phone system or even
worse, service from the "phone company", that requires dialing 9 to get an "outside line".

I bet even the police dept has to dial 9 to dial 911.

And as far as that goes, put the line in your pet tarantula's name, and make it clear to them
that the spider is more then willing to 'serve its time' for such a terrible transgressions!! :-)

Wholesale outbound telecom services, such as ones provided by most SIP providers, are *NOT* 
"in your name", *NOR* do they even technically have a "phone number" or a "service address".
Just because you pay the bill for them does not, in any manner, construe that you are the end user
of the service.

Jim




Date: Thu, 5 Sep 2013 14:58:42 -0400
Subject: RE: [App_rpt-users] SIP VoIP for Asterisk
From: DwaineGarden at rogers.com
To: telesistant at hotmail.com
CC: dshaw at ke6upi.com; app_rpt-users at ohnosec.org

It was an interesting experience.  Has anyone else experienced the same problem and had the police involved?  
I did mention that I would change the
Dial string.  The police told me that a person has to be able to dial 911 and get emergency services.

I gave up and just walked over to the all-star computer and hit the power button.  Sadly, it has been off ever since.

If I had changed the 911 Dial string before the incident.  The police would of not gotten involved. Never known about the situation.

Just let everyone know.  The two police officers were very good about it.  They told me that its happening a lot and 911 is getting a little upset about it.

I told them they should go after the person responsible.  They told me that's you.   The line is in your name.

Anyway, just wanted to throw out there my experience.

Dwaine

Jim Duuuude <telesistant at hotmail.com> wrote:

Okay (and yes, that is STUPID and most likely WRONG, but most police depts are
just completely ignorant of telecom issues). So, fine... GIVE them access to 911.
Let them dial it. But, sadly, on YOUR phone network, the dialing string is just a LITTLE
bit longer (like about 30 digits in front of the 911)... get it?

Jim


Date: Thu, 5 Sep 2013 14:07:58 -0400
From: DwaineGarden at rogers.com
To: dshaw at ke6upi.com
CC: app_rpt-users at ohnosec.org
Subject: Re: [App_rpt-users] SIP VoIP for Asterisk

The problem is you are not allowed by law to have a phone without unrestricted access to 911.  I had Metro Toronto police at my door explaining that even if I block 911 to any outside connections I would be breaking the law.  If you have a server on the internet with sip.  They have to able to connect to be able to call 911. 

I told the police it was retard.  They told me that was fine they will charge me.

Police told me that even if someone breaks into your house.  If there is a phone install,  the criminals better have access to dial 911 unrestrictive.

The hackers did not get into the box.  They were trying for months.   Got pissed off and changed their script to dial 911 constantly.  SIP and DID see a 911 call.  It dials it.  No questions asked.  No login or nothing.

The Police told me it was a huge problem.  SIP or DID are setup like a public pay phone.  Full access to 911.

It was an eye opener for me.   You learn something new everyday.  If I see someone asking about SIP or DID.  I let them know about my experience.

David KE6UPI <dshaw at ke6upi.com> wrote:

I'm sorry Dwaine what are you talking about? Sorry If I don't understand what your talking about.

I have both a public Asterisk server and a local Asterisk server. I have never had anyone connect and make a call that was not authenticated user.. Yes they try and fail2ban will block them. There are many way to stop unwanted hackers on your server. 



As for dialing 911 just make a dial plain to route to space if you want. 

Google "Asterisk Security"

http://www.voip-info.org/wiki/view/Asterisk+security



David



On Thu, Sep 5, 2013 at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:

It works great until the hacks find the machine.  They port scan non stop.  Its especially fun when their scripts dial 911 constantly.  There is no way to turn off dialing 911 for SIP.




Bill South <wbs099 at yahoo.com> wrote:



>     I'm thinking of adding some type of SIP trunking or other VoIP service provider to my ACID Asterisk system to support in/out bound calling.  I've read some emails on the app_rpt reflector with names of providers mentioned, but I am looking for recommendations, as there are zillions of VoIP providers out there.  This is going to be used very sparingly so least-cost is a good thing, but good reliability and no bombardment with email adds by the provider is desired too.  I can easily get by with a single number, but may want to add additional DIDs later.  Thoughts?


>

>Bill

>

>

>_______________________________________________

>App_rpt-users mailing list

>App_rpt-users at ohnosec.org

>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

_______________________________________________

App_rpt-users mailing list

App_rpt-users at ohnosec.org

http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users




_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 		 	   		   		 	   		  
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From DwaineGarden at rogers.com  Thu Sep  5 19:40:32 2013
From: DwaineGarden at rogers.com (Dwaine Garden VE3GIF)
Date: Thu, 05 Sep 2013 15:40:32 -0400
Subject: [App_rpt-users] SIP VoIP for Asterisk
Message-ID: <dc6y8cvjn65q0cgcm6336up6.1378410032252@email.android.com>

Canadian customers have to provide an address and name for 911 purposes for a DID.  Bell and Rogers were mandated with their VoIP services.

I gave them the logs of the server where the IP address were listed.   Rogers was good and moved me to a new IP address and filtered out the source.

I had an iptables script that would ban IP address that attempted a certain number of connections during a certain amount of time.  Then ban that IP address.  It worked great.  It stopped the hackers script dead cold.  Until they ran the 911 script.  They were ruthless in their attacks.

My all-star node was the best fun I have had with Ham radio.  I'll fire it back up again.   Just waiting for the dust to settle without a DID setup.

Dwaine

Jim Duuuude <telesistant at hotmail.com> wrote:

><!-- .hmmessage P { margin:0px; padding:0px } body.hmmessage { font-size: 12pt; font-family:Calibri } -->
>
>Double BRAVO-SIERRA!!
>
>If that were true, a LOT of large business owners (some of which may even operate
>large businesses :-) ) would be in jail, if they HAPPEN to have a phone system or even
>worse, service from the "phone company", that requires dialing 9 to get an "outside line".
>
>I bet even the police dept has to dial 9 to dial 911.
>
>And as far as that goes, put the line in your pet tarantula's name, and make it clear to them
>that the spider is more then willing to 'serve its time' for such a terrible transgressions!! :-)
>
>Wholesale outbound telecom services, such as ones provided by most SIP providers, are *NOT* 
>"in your name", *NOR* do they even technically have a "phone number" or a "service address".
>Just because you pay the bill for them does not, in any manner, construe that you are the end user
>of the service.
>
>Jim
>
>
>
>
>Date: Thu, 5 Sep 2013 14:58:42 -0400
>Subject: RE: [App_rpt-users] SIP VoIP for Asterisk
>From: DwaineGarden at rogers.com
>To: telesistant at hotmail.com
>CC: dshaw at ke6upi.com; app_rpt-users at ohnosec.org
>
>It was an interesting experience. Has anyone else experienced the same problem and had the police involved? 
>I did mention that I would change the
>Dial string. The police told me that a person has to be able to dial 911 and get emergency services.
>
>I gave up and just walked over to the all-star computer and hit the power button. Sadly, it has been off ever since.
>
>If I had changed the 911 Dial string before the incident. The police would of not gotten involved. Never known about the situation.
>
>Just let everyone know. The two police officers were very good about it. They told me that its happening a lot and 911 is getting a little upset about it.
>
>I told them they should go after the person responsible. They told me that's you. The line is in your name.
>
>Anyway, just wanted to throw out there my experience.
>
>Dwaine
>
>Jim Duuuude <telesistant at hotmail.com> wrote:
>
>Okay (and yes, that is STUPID and most likely WRONG, but most police depts are
>just completely ignorant of telecom issues). So, fine... GIVE them access to 911.
>Let them dial it. But, sadly, on YOUR phone network, the dialing string is just a LITTLE
>bit longer (like about 30 digits in front of the 911)... get it?
>
>Jim
>
>
>Date: Thu, 5 Sep 2013 14:07:58 -0400
>From: DwaineGarden at rogers.com
>To: dshaw at ke6upi.com
>CC: app_rpt-users at ohnosec.org
>Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
>
>The problem is you are not allowed by law to have a phone without unrestricted access to 911. I had Metro Toronto police at my door explaining that even if I block 911 to any outside connections I would be breaking the law. If you have a server on the internet with sip. They have to able to connect to be able to call 911. 
>
>I told the police it was retard. They told me that was fine they will charge me.
>
>Police told me that even if someone breaks into your house. If there is a phone install, the criminals better have access to dial 911 unrestrictive.
>
>The hackers did not get into the box. They were trying for months. Got pissed off and changed their script to dial 911 constantly. SIP and DID see a 911 call. It dials it. No questions asked. No login or nothing.
>
>The Police told me it was a huge problem. SIP or DID are setup like a public pay phone. Full access to 911.
>
>It was an eye opener for me. You learn something new everyday. If I see someone asking about SIP or DID. I let them know about my experience.
>
>David KE6UPI <dshaw at ke6upi.com> wrote:
>
>I'm sorry Dwaine what are you talking about? Sorry If I don't understand what your talking about.
>
>I have both a public Asterisk server and a local Asterisk server. I have never had anyone connect and make a call that was not authenticated user.. Yes they try and fail2ban will block them. There are many way to stop unwanted hackers on your server. 
>
>
>As for dialing 911 just make a dial plain to route to space if you want. 
>
>Google "Asterisk Security"
>
>
>http://www.voip-info.org/wiki/view/Asterisk+security
>
>
>David
>
>
>
>
>On Thu, Sep 5, 2013 at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:
>
>It works great until the hacks find the machine. ?They port scan non stop. ?Its especially fun when their scripts dial 911 constantly. ?There is no way to turn off dialing 911 for SIP.
>
>
>Bill South <wbs099 at yahoo.com> wrote:
>
>> ? ? I'm thinking of adding some type of SIP trunking or other VoIP service provider to my ACID Asterisk system to support in/out bound calling. ?I've read some emails on the app_rpt reflector with names of providers mentioned, but I am looking for recommendations, as there are zillions of VoIP providers out there. ?This is going to be used very sparingly so least-cost is a good thing, but good reliability and no bombardment with email adds by the provider is desired too. ?I can easily get by with a single number, but may want to add additional DIDs later. ?Thoughts?
>>
>>Bill
>>
>>
>>_______________________________________________
>>App_rpt-users mailing list
>>App_rpt-users at ohnosec.org
>>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
>
>_______________________________________________ App_rpt-users mailing list App_rpt-users at ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>		 	 		 
>
>		 	 		 
>
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From wbs099 at yahoo.com  Thu Sep  5 20:39:11 2013
From: wbs099 at yahoo.com (Bill South)
Date: Thu, 5 Sep 2013 13:39:11 -0700 (PDT)
Subject: [App_rpt-users] SIP VoIP for Asterisk
In-Reply-To: <dc6y8cvjn65q0cgcm6336up6.1378410032252@email.android.com>
Message-ID: <1378413551.80356.YahooMailBasic@web161405.mail.bf1.yahoo.com>

     While this thread is not part of what I originally asked about (SIP provider recommendations) I'll add that 911 access is mandated in some LATAs, but seems to be some debate if it must be 911 or if 9-911 is OK, or other variations.  I worked in the telecom business for over 25 years in various parts of the USA and different localities would respond differently to the question about 911.  When E911 first came out, where a database lookup was done by the local police emeregency agency systems answering 911 calls showing names and addresses, some jurisdictions mandated that database had to show the exact location the call was originating from, not a billing address of some companies headquarters.  Big problem for some large companies where a single billing address is used for all telco circuits.  For residences I'm not sure anyone has put much thought into laws regarding 911 dialing and any restrictions thereof; businesses on the other hand, where
 100's maybe thousands of workers are in the same building and floor and locating a 911 caller could be pretty tough for local emnergency responders, there are, or were anyway, laws in some locations mandating 911 unrestricted.  The thought has always been though that in an emergency people have the expectation, whether at home or work, that they can dial 911 and get help; not sure where the laws stands now on that across the USA.

--------------------------------------------
On Thu, 9/5/13, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:

 Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
 To: "Jim Duuuude" <telesistant at hotmail.com>
 Cc: "app_rpt mailing list" <app_rpt-users at ohnosec.org>
 Date: Thursday, September 5, 2013, 7:40 PM
 
 Canadian customers have to provide an
 address and name for 911 purposes for a DID.  Bell and
 Rogers were mandated with their VoIP services.
 
 I gave them the logs of the server where the IP address were
 listed.   Rogers was good and moved me to a new IP address
 and filtered out the source.
 
 I had an iptables script that would ban IP address that
 attempted a certain number of connections during a certain
 amount of time.  Then ban that IP address.  It worked great.
  It stopped the hackers script dead cold.  Until they ran
 the 911 script.  They were ruthless in their attacks.
 
 My all-star node was the best fun I have had with Ham radio.
  I'll fire it back up again.   Just waiting for the dust
 to settle without a DID setup.
 
 Dwaine
 
 Jim Duuuude <telesistant at hotmail.com> wrote:
 
 Double BRAVO-SIERRA!!
 
 If that were true, a LOT of large business owners (some of
 which may even operate
 large businesses :-) ) would be in jail, if they HAPPEN to
 have a phone system or even
 worse, service from the "phone company", that
 requires dialing 9 to get an "outside line".
 
 I bet even the police dept has to dial 9 to dial 911.
 
 And as far as that goes, put the line in your pet
 tarantula's name, and make it clear to them
 that the spider is more then willing to 'serve its
 time' for such a terrible transgressions!! :-)
 
 Wholesale outbound telecom services, such as ones provided
 by most SIP providers, are *NOT* 
 "in your name", *NOR* do they even technically
 have a "phone number" or a "service
 address".
 Just because you pay the bill for them does not, in any
 manner, construe that you are the end user
 of the service.
 
 Jim
 
 
 
 
 Date: Thu, 5 Sep
 2013 14:58:42 -0400
 Subject: RE: [App_rpt-users] SIP VoIP for Asterisk
 From: DwaineGarden at rogers.com
 To: telesistant at hotmail.com
 CC: dshaw at ke6upi.com; app_rpt-users at ohnosec.org
 
 It was an interesting experience.  Has anyone else
 experienced the same problem and had the police involved?  
 I did mention that I would change the
 Dial string.  The police told me that a person has to be
 able to dial 911 and get emergency services.
 
 I gave up and just walked over to the all-star computer and
 hit the power button.  Sadly, it has been off ever since.
 
 If I had changed the 911 Dial string before the incident. 
 The police would of not gotten involved. Never known about
 the situation.
 
 Just let everyone know.  The two police officers were very
 good about it.  They told me that its happening a lot and
 911 is getting a little upset about it.
 
 I told them they should go after the person responsible. 
 They told me that's you.   The line is in your name.
 
 Anyway, just wanted to throw out there my experience.
 
 Dwaine
 
 Jim Duuuude <telesistant at hotmail.com> wrote:
 
 Okay (and yes, that is STUPID and most likely
 WRONG, but most police depts are
 just completely ignorant of telecom issues). So, fine...
 GIVE them access to 911.
 Let them dial it. But, sadly, on YOUR phone network, the
 dialing string is just a LITTLE
 bit longer (like about 30 digits in front of the 911)... get
 it?
 
 Jim
 
 
 Date: Thu, 5 Sep
 2013 14:07:58 -0400
 From: DwaineGarden at rogers.com
 To: dshaw at ke6upi.com
 CC: app_rpt-users at ohnosec.org
 Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
 
 The problem is you are not allowed by law to have a phone
 without unrestricted access to 911.  I had Metro Toronto
 police at my door explaining that even if I block 911 to any
 outside connections I would be breaking the law.  If you
 have a server on the internet with sip.  They have to able
 to connect to be able to call 911. 
 
 I told the police it was retard.  They told me that was fine
 they will charge me.
 
 Police told me that even if someone breaks into your house. 
 If there is a phone install,  the criminals better have
 access to dial 911 unrestrictive.
 
 The hackers did not get into the box.  They were trying for
 months.   Got pissed off and changed their script to dial
 911 constantly.  SIP and DID see a 911 call.  It dials it. 
 No questions asked.  No login or nothing.
 
 The Police told me it was a huge problem.  SIP or DID are
 setup like a public pay phone.  Full access to 911.
 
 It was an eye opener for me.   You learn something new
 everyday.  If I see someone asking about SIP or DID.  I let
 them know about my experience.
 
 David KE6UPI <dshaw at ke6upi.com> wrote:
 
 I'm sorry Dwaine what are you
 talking about? Sorry If I don't understand what your
 talking about.
 
 I have both a public Asterisk server and a local
 Asterisk server. I have never had anyone connect and make a
 call that was not authenticated user.. Yes they try and
 fail2ban will block them. There are many way to stop
 unwanted hackers on your server. 
 
 
 
 As for dialing 911 just make a dial plain to route to space
 if you want. 
 
 Google "Asterisk Security"
 
 http://www.voip-info.org/wiki/view/Asterisk+security
 
 
 
 David
 
 
 
 On Thu, Sep 5, 2013
 at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com>
 wrote:
 
 It
 works great until the hacks find the machine. ?They
 port scan non stop. ?Its especially fun when their
 scripts dial 911 constantly. ?There is no way to turn
 off dialing 911 for SIP.
 
 
 
 
 Bill South <wbs099 at yahoo.com>
 wrote:
 
 
 
 > ? ? I'm thinking of adding some type of
 SIP trunking or other VoIP service provider to my ACID
 Asterisk system to support in/out bound calling.
 ?I've read some emails on the app_rpt reflector
 with names of providers mentioned, but I am looking for
 recommendations, as there are zillions of VoIP providers out
 there. ?This is going to be used very sparingly so
 least-cost is a good thing, but good reliability and no
 bombardment with email adds by the provider is desired too.
 ?I can easily get by with a single number, but may want
 to add additional DIDs later. ?Thoughts?
 
 
 >
 
 >Bill
 
 >
 
 >
 
 >_______________________________________________
 
 >App_rpt-users mailing list
 
 >App_rpt-users at ohnosec.org
 
 >http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
 
 _______________________________________________
 
 App_rpt-users mailing list
 
 App_rpt-users at ohnosec.org
 
 http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
 
 
 
 
 _______________________________________________
 App_rpt-users mailing list
 App_rpt-users at ohnosec.org
 http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
 		 	   		   		 	   		  
 -----Inline Attachment Follows-----
 
 _______________________________________________
 App_rpt-users mailing list
 App_rpt-users at ohnosec.org
 http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
 


From DwaineGarden at rogers.com  Thu Sep  5 22:29:18 2013
From: DwaineGarden at rogers.com (Dwaine Garden VE3GIF)
Date: Thu, 05 Sep 2013 18:29:18 -0400
Subject: [App_rpt-users] SIP VoIP for Asterisk
Message-ID: <7phw1oohlkk6yllownxqc1wa.1378419993049@email.android.com>

I walked over to the telco team where I work and mentioned our conversation to them.   They told me that the law states you have to clearly label the telephone device if you change the dialup number from 911.   (That's why we have all our VOIP phones labeled with the different number)

I mentioned the AllStar node.   They said E911 is a complete mess.  You can't label an outside connection to a VOIP server.  You would not want to either if you could.

Most companies are just ignoring the requirement to have an open dial plan for E911 from the Internet.  If you find yourself in a legal situation.   The telco guys told me your at fault.  You get fined.

E911 should really be traceable back to the person that placed the call.  They can do that with certain, so the law goes after the person which e911 is registered under.

The guys in the telco team just laughed when I talked to them about the AllStar node.   Welcome to VOIP and the E911 mess.

Dwaine

Bill South <wbs099 at yahoo.com> wrote:

>     While this thread is not part of what I originally asked about (SIP provider recommendations) I'll add that 911 access is mandated in some LATAs, but seems to be some debate if it must be 911 or if 9-911 is OK, or other variations.  I worked in the telecom business for over 25 years in various parts of the USA and different localities would respond differently to the question about 911.  When E911 first came out, where a database lookup was done by the local police emeregency agency systems answering 911 calls showing names and addresses, some jurisdictions mandated that database had to show the exact location the call was originating from, not a billing address of some companies headquarters.  Big problem for some large companies where a single billing address is used for all telco circuits.  For residences I'm not sure anyone has put much thought into laws regarding 911 dialing and any restrictions thereof; businesses on the other hand, where
> 100's maybe thousands of workers are in the same building and floor and locating a 911 caller could be pretty tough for local emnergency responders, there are, or were anyway, laws in some locations mandating 911 unrestricted.  The thought has always been though that in an emergency people have the expectation, whether at home or work, that they can dial 911 and get help; not sure where the laws stands now on that across the USA.
>
>--------------------------------------------
>On Thu, 9/5/13, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:
>
> Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
> To: "Jim Duuuude" <telesistant at hotmail.com>
> Cc: "app_rpt mailing list" <app_rpt-users at ohnosec.org>
> Date: Thursday, September 5, 2013, 7:40 PM
> 
> Canadian customers have to provide an
> address and name for 911 purposes for a DID.  Bell and
> Rogers were mandated with their VoIP services.
> 
> I gave them the logs of the server where the IP address were
> listed.   Rogers was good and moved me to a new IP address
> and filtered out the source.
> 
> I had an iptables script that would ban IP address that
> attempted a certain number of connections during a certain
> amount of time.  Then ban that IP address.  It worked great.
>  It stopped the hackers script dead cold.  Until they ran
> the 911 script.  They were ruthless in their attacks.
> 
> My all-star node was the best fun I have had with Ham radio.
>  I'll fire it back up again.   Just waiting for the dust
> to settle without a DID setup.
> 
> Dwaine
> 
> Jim Duuuude <telesistant at hotmail.com> wrote:
> 
> Double BRAVO-SIERRA!!
> 
> If that were true, a LOT of large business owners (some of
> which may even operate
> large businesses :-) ) would be in jail, if they HAPPEN to
> have a phone system or even
> worse, service from the "phone company", that
> requires dialing 9 to get an "outside line".
> 
> I bet even the police dept has to dial 9 to dial 911.
> 
> And as far as that goes, put the line in your pet
> tarantula's name, and make it clear to them
> that the spider is more then willing to 'serve its
> time' for such a terrible transgressions!! :-)
> 
> Wholesale outbound telecom services, such as ones provided
> by most SIP providers, are *NOT* 
> "in your name", *NOR* do they even technically
> have a "phone number" or a "service
> address".
> Just because you pay the bill for them does not, in any
> manner, construe that you are the end user
> of the service.
> 
> Jim
> 
> 
> 
> 
> Date: Thu, 5 Sep
> 2013 14:58:42 -0400
> Subject: RE: [App_rpt-users] SIP VoIP for Asterisk
> From: DwaineGarden at rogers.com
> To: telesistant at hotmail.com
> CC: dshaw at ke6upi.com; app_rpt-users at ohnosec.org
> 
> It was an interesting experience.  Has anyone else
> experienced the same problem and had the police involved?  
> I did mention that I would change the
> Dial string.  The police told me that a person has to be
> able to dial 911 and get emergency services.
> 
> I gave up and just walked over to the all-star computer and
> hit the power button.  Sadly, it has been off ever since.
> 
> If I had changed the 911 Dial string before the incident. 
> The police would of not gotten involved. Never known about
> the situation.
> 
> Just let everyone know.  The two police officers were very
> good about it.  They told me that its happening a lot and
> 911 is getting a little upset about it.
> 
> I told them they should go after the person responsible. 
> They told me that's you.   The line is in your name.
> 
> Anyway, just wanted to throw out there my experience.
> 
> Dwaine
> 
> Jim Duuuude <telesistant at hotmail.com> wrote:
> 
> Okay (and yes, that is STUPID and most likely
> WRONG, but most police depts are
> just completely ignorant of telecom issues). So, fine...
> GIVE them access to 911.
> Let them dial it. But, sadly, on YOUR phone network, the
> dialing string is just a LITTLE
> bit longer (like about 30 digits in front of the 911)... get
> it?
> 
> Jim
> 
> 
> Date: Thu, 5 Sep
> 2013 14:07:58 -0400
> From: DwaineGarden at rogers.com
> To: dshaw at ke6upi.com
> CC: app_rpt-users at ohnosec.org
> Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
> 
> The problem is you are not allowed by law to have a phone
> without unrestricted access to 911.  I had Metro Toronto
> police at my door explaining that even if I block 911 to any
> outside connections I would be breaking the law.  If you
> have a server on the internet with sip.  They have to able
> to connect to be able to call 911. 
> 
> I told the police it was retard.  They told me that was fine
> they will charge me.
> 
> Police told me that even if someone breaks into your house. 
> If there is a phone install,  the criminals better have
> access to dial 911 unrestrictive.
> 
> The hackers did not get into the box.  They were trying for
> months.   Got pissed off and changed their script to dial
> 911 constantly.  SIP and DID see a 911 call.  It dials it. 
> No questions asked.  No login or nothing.
> 
> The Police told me it was a huge problem.  SIP or DID are
> setup like a public pay phone.  Full access to 911.
> 
> It was an eye opener for me.   You learn something new
> everyday.  If I see someone asking about SIP or DID.  I let
> them know about my experience.
> 
> David KE6UPI <dshaw at ke6upi.com> wrote:
> 
> I'm sorry Dwaine what are you
> talking about? Sorry If I don't understand what your
> talking about.
> 
> I have both a public Asterisk server and a local
> Asterisk server. I have never had anyone connect and make a
> call that was not authenticated user.. Yes they try and
> fail2ban will block them. There are many way to stop
> unwanted hackers on your server. 
> 
> 
> 
> As for dialing 911 just make a dial plain to route to space
> if you want. 
> 
> Google "Asterisk Security"
> 
> http://www.voip-info.org/wiki/view/Asterisk+security
> 
> 
> 
> David
> 
> 
> 
> On Thu, Sep 5, 2013
> at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com>
> wrote:
> 
> It
> works great until the hacks find the machine. ?They
> port scan non stop. ?Its especially fun when their
> scripts dial 911 constantly. ?There is no way to turn
> off dialing 911 for SIP.
> 
> 
> 
> 
> Bill South <wbs099 at yahoo.com>
> wrote:
> 
> 
> 
> > ? ? I'm thinking of adding some type of
> SIP trunking or other VoIP service provider to my ACID
> Asterisk system to support in/out bound calling.
> ?I've read some emails on the app_rpt reflector
> with names of providers mentioned, but I am looking for
> recommendations, as there are zillions of VoIP providers out
> there. ?This is going to be used very sparingly so
> least-cost is a good thing, but good reliability and no
> bombardment with email adds by the provider is desired too.
> ?I can easily get by with a single number, but may want
> to add additional DIDs later. ?Thoughts?
> 
> 
> >
> 
> >Bill
> 
> >
> 
> >
> 
> >_______________________________________________
> 
> >App_rpt-users mailing list
> 
> >App_rpt-users at ohnosec.org
> 
> >http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> _______________________________________________
> 
> App_rpt-users mailing list
> 
> App_rpt-users at ohnosec.org
> 
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> 
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 		 	   		   		 	   		  
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

From DwaineGarden at rogers.com  Thu Sep  5 22:32:24 2013
From: DwaineGarden at rogers.com (Dwaine Garden VE3GIF)
Date: Thu, 05 Sep 2013 18:32:24 -0400
Subject: [App_rpt-users] SIP VoIP for Asterisk
Message-ID: <ut8ketm5irym5r4lf0kx7fw1.1378420344915@email.android.com>

Bill, sorry for hijacking your thread.  Didn't mean it....

Dwine

Bill South <wbs099 at yahoo.com> wrote:

>     While this thread is not part of what I originally asked about (SIP provider recommendations) I'll add that 911 access is mandated in some LATAs, but seems to be some debate if it must be 911 or if 9-911 is OK, or other variations.  I worked in the telecom business for over 25 years in various parts of the USA and different localities would respond differently to the question about 911.  When E911 first came out, where a database lookup was done by the local police emeregency agency systems answering 911 calls showing names and addresses, some jurisdictions mandated that database had to show the exact location the call was originating from, not a billing address of some companies headquarters.  Big problem for some large companies where a single billing address is used for all telco circuits.  For residences I'm not sure anyone has put much thought into laws regarding 911 dialing and any restrictions thereof; businesses on the other hand, where
> 100's maybe thousands of workers are in the same building and floor and locating a 911 caller could be pretty tough for local emnergency responders, there are, or were anyway, laws in some locations mandating 911 unrestricted.  The thought has always been though that in an emergency people have the expectation, whether at home or work, that they can dial 911 and get help; not sure where the laws stands now on that across the USA.
>
>--------------------------------------------
>On Thu, 9/5/13, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:
>
> Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
> To: "Jim Duuuude" <telesistant at hotmail.com>
> Cc: "app_rpt mailing list" <app_rpt-users at ohnosec.org>
> Date: Thursday, September 5, 2013, 7:40 PM
> 
> Canadian customers have to provide an
> address and name for 911 purposes for a DID.  Bell and
> Rogers were mandated with their VoIP services.
> 
> I gave them the logs of the server where the IP address were
> listed.   Rogers was good and moved me to a new IP address
> and filtered out the source.
> 
> I had an iptables script that would ban IP address that
> attempted a certain number of connections during a certain
> amount of time.  Then ban that IP address.  It worked great.
>  It stopped the hackers script dead cold.  Until they ran
> the 911 script.  They were ruthless in their attacks.
> 
> My all-star node was the best fun I have had with Ham radio.
>  I'll fire it back up again.   Just waiting for the dust
> to settle without a DID setup.
> 
> Dwaine
> 
> Jim Duuuude <telesistant at hotmail.com> wrote:
> 
> Double BRAVO-SIERRA!!
> 
> If that were true, a LOT of large business owners (some of
> which may even operate
> large businesses :-) ) would be in jail, if they HAPPEN to
> have a phone system or even
> worse, service from the "phone company", that
> requires dialing 9 to get an "outside line".
> 
> I bet even the police dept has to dial 9 to dial 911.
> 
> And as far as that goes, put the line in your pet
> tarantula's name, and make it clear to them
> that the spider is more then willing to 'serve its
> time' for such a terrible transgressions!! :-)
> 
> Wholesale outbound telecom services, such as ones provided
> by most SIP providers, are *NOT* 
> "in your name", *NOR* do they even technically
> have a "phone number" or a "service
> address".
> Just because you pay the bill for them does not, in any
> manner, construe that you are the end user
> of the service.
> 
> Jim
> 
> 
> 
> 
> Date: Thu, 5 Sep
> 2013 14:58:42 -0400
> Subject: RE: [App_rpt-users] SIP VoIP for Asterisk
> From: DwaineGarden at rogers.com
> To: telesistant at hotmail.com
> CC: dshaw at ke6upi.com; app_rpt-users at ohnosec.org
> 
> It was an interesting experience.  Has anyone else
> experienced the same problem and had the police involved?  
> I did mention that I would change the
> Dial string.  The police told me that a person has to be
> able to dial 911 and get emergency services.
> 
> I gave up and just walked over to the all-star computer and
> hit the power button.  Sadly, it has been off ever since.
> 
> If I had changed the 911 Dial string before the incident. 
> The police would of not gotten involved. Never known about
> the situation.
> 
> Just let everyone know.  The two police officers were very
> good about it.  They told me that its happening a lot and
> 911 is getting a little upset about it.
> 
> I told them they should go after the person responsible. 
> They told me that's you.   The line is in your name.
> 
> Anyway, just wanted to throw out there my experience.
> 
> Dwaine
> 
> Jim Duuuude <telesistant at hotmail.com> wrote:
> 
> Okay (and yes, that is STUPID and most likely
> WRONG, but most police depts are
> just completely ignorant of telecom issues). So, fine...
> GIVE them access to 911.
> Let them dial it. But, sadly, on YOUR phone network, the
> dialing string is just a LITTLE
> bit longer (like about 30 digits in front of the 911)... get
> it?
> 
> Jim
> 
> 
> Date: Thu, 5 Sep
> 2013 14:07:58 -0400
> From: DwaineGarden at rogers.com
> To: dshaw at ke6upi.com
> CC: app_rpt-users at ohnosec.org
> Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
> 
> The problem is you are not allowed by law to have a phone
> without unrestricted access to 911.  I had Metro Toronto
> police at my door explaining that even if I block 911 to any
> outside connections I would be breaking the law.  If you
> have a server on the internet with sip.  They have to able
> to connect to be able to call 911. 
> 
> I told the police it was retard.  They told me that was fine
> they will charge me.
> 
> Police told me that even if someone breaks into your house. 
> If there is a phone install,  the criminals better have
> access to dial 911 unrestrictive.
> 
> The hackers did not get into the box.  They were trying for
> months.   Got pissed off and changed their script to dial
> 911 constantly.  SIP and DID see a 911 call.  It dials it. 
> No questions asked.  No login or nothing.
> 
> The Police told me it was a huge problem.  SIP or DID are
> setup like a public pay phone.  Full access to 911.
> 
> It was an eye opener for me.   You learn something new
> everyday.  If I see someone asking about SIP or DID.  I let
> them know about my experience.
> 
> David KE6UPI <dshaw at ke6upi.com> wrote:
> 
> I'm sorry Dwaine what are you
> talking about? Sorry If I don't understand what your
> talking about.
> 
> I have both a public Asterisk server and a local
> Asterisk server. I have never had anyone connect and make a
> call that was not authenticated user.. Yes they try and
> fail2ban will block them. There are many way to stop
> unwanted hackers on your server. 
> 
> 
> 
> As for dialing 911 just make a dial plain to route to space
> if you want. 
> 
> Google "Asterisk Security"
> 
> http://www.voip-info.org/wiki/view/Asterisk+security
> 
> 
> 
> David
> 
> 
> 
> On Thu, Sep 5, 2013
> at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com>
> wrote:
> 
> It
> works great until the hacks find the machine. ?They
> port scan non stop. ?Its especially fun when their
> scripts dial 911 constantly. ?There is no way to turn
> off dialing 911 for SIP.
> 
> 
> 
> 
> Bill South <wbs099 at yahoo.com>
> wrote:
> 
> 
> 
> > ? ? I'm thinking of adding some type of
> SIP trunking or other VoIP service provider to my ACID
> Asterisk system to support in/out bound calling.
> ?I've read some emails on the app_rpt reflector
> with names of providers mentioned, but I am looking for
> recommendations, as there are zillions of VoIP providers out
> there. ?This is going to be used very sparingly so
> least-cost is a good thing, but good reliability and no
> bombardment with email adds by the provider is desired too.
> ?I can easily get by with a single number, but may want
> to add additional DIDs later. ?Thoughts?
> 
> 
> >
> 
> >Bill
> 
> >
> 
> >
> 
> >_______________________________________________
> 
> >App_rpt-users mailing list
> 
> >App_rpt-users at ohnosec.org
> 
> >http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> _______________________________________________
> 
> App_rpt-users mailing list
> 
> App_rpt-users at ohnosec.org
> 
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> 
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 		 	   		   		 	   		  
> -----Inline Attachment Follows-----
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
>_______________________________________________
>App_rpt-users mailing list
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>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

From tim.sawyer at me.com  Thu Sep  5 23:58:17 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Thu, 05 Sep 2013 16:58:17 -0700
Subject: [App_rpt-users] permit=  echolink.conf
In-Reply-To: <bf04a.50e55d88.3f5a2697@aol.com>
References: <bf04a.50e55d88.3f5a2697@aol.com>
Message-ID: <779F66B2-2D01-49E2-B3E4-B48376031A95@me.com>

Look at /usr/src/astsrc/asterisk/channels/chan_echolink.c

I'm no good at C but I have a stinking suspicion that increasing EL_MAX_CALL_LIST would allow both bigger permit and deny lists :-)
--
Tim
:wq

On Sep 5, 2013, at 11:25 AM, N7LD at aol.com wrote:

> Looks like I am having the same problem. Looks like the MAX stations is 30
> Lee
>  
>  
> In a message dated 5/21/2013 11:08:47 A.M. Pacific Daylight Time, w8jtw at yahoo.com writes:
> Is there anyway to increase the permit=  list in echolink.conf to work with more than 25 or so stations?  Once I get beyond 25 it does not want to allow beyond the first 25.  Or is there a way to limit to just USA connections? 
>     I dont want to run a wide open echolink but have many friends i don't want to block    need to have capacity of about 100 or so.
> 
> Thought of trying Permit=W*,N*,A*,K* not sure if that would limit to just US callsigns or not.
> 
> Any suggestions?
> 
> Thanks
> 
> Joe 
> W8JTW
> Node 27891  
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From N7LD at aol.com  Fri Sep  6 00:43:44 2013
From: N7LD at aol.com (N7LD at aol.com)
Date: Thu, 5 Sep 2013 20:43:44 -0400 (EDT)
Subject: [App_rpt-users] permit=  echolink.conf
Message-ID: <e8569.64b88359.3f5a7f40@aol.com>

Thanks Tim, that worked!!!
Set mine to 60 and it works.
Lee
 
 
 
In a message dated 9/5/2013 4:58:21 P.M. Pacific Daylight Time,  
tim.sawyer at me.com writes:

Look at /usr/src/astsrc/asterisk/channels/chan_echolink.c


I'm no good at C but I have a stinking suspicion that  increasing 
EL_MAX_CALL_LIST would allow both bigger permit and deny lists  :-)
 
--
Tim
:wq



On Sep 5, 2013, at 11:25 AM, _N7LD at aol.com_ (mailto:N7LD at aol.com)  wrote:



Looks like I am having the same problem. Looks like the MAX stations is  30
Lee
 
 
 
In a message dated 5/21/2013 11:08:47 A.M. Pacific Daylight Time, 
_w8jtw at yahoo.com_ (mailto:w8jtw at yahoo.com)  writes:

Is there anyway to  increase the permit=  list in echolink.conf to work 
with more than 25  or so stations?  Once I get beyond 25 it does not want to 
allow  beyond the first 25.  Or is there a way to limit to just USA  
connections? 
I dont want to run a wide open echolink but  have many friends i don't want 
to block    need to have capacity  of about 100 or so.

Thought of trying Permit=W*,N*,A*,K* not sure  if that would limit to just 
US callsigns or not.

Any  suggestions?

Thanks

Joe 
W8JTW
Node 27891   
_______________________________________________
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_App_rpt-users at ohnosec.org_ (mailto:App_rpt-users at ohnosec.org) 
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From ve3elb at yahoo.com  Fri Sep  6 01:12:26 2013
From: ve3elb at yahoo.com (Vince P)
Date: Thu, 5 Sep 2013 21:12:26 -0400
Subject: [App_rpt-users] Setting System Time
In-Reply-To: <1378354706.75967.YahooMailNeo@web163602.mail.gq1.yahoo.com>
References: <CD88227D-950D-4233-AD99-673A1670AC31@yahoo.com>
	<1378354706.75967.YahooMailNeo@web163602.mail.gq1.yahoo.com>
Message-ID: <F6815E65-97A0-45F6-87B2-7DC7ACB5F880@yahoo.com>

Thank You to everyone for your replies and info. 

Thank You Johnny your instructions work perfectly. 



Vince VE3ELB




On 2013-09-05, at 12:18 AM, Johnny Keeker <cypresstower at yahoo.com> wrote:

Set up Timezone Carfully Copy each line below to your linux box as root. 
Before begining, check /usr/share/zoneinfo/ for your time zone.
Enter your time zone, a space, followed by the word localtime
The example below is if you lived in Toronto Canada.
 
[root at node /]#cd /etc
[root at node /]#cp localtime localtime.orig
[root at node /]#ln ?-sf /usr/share/zoneinfo/Canada/Eastern localtime
 
Check your results
[root at node /]#date
 
Synchronize time from the web
[root at node /]#ntpdate pool.ntp.org
 
Set Server to Auto Sync The Time from the web
[root at node /]#ntpdate 0.us.pool.ntp.org
[root at node /]#hwclock --systohc
 
Restart ntpd service
[root at node /]#service ntpd start
 
Enable service at startup
[root at node /]#chkconfig ntpd on

 
From: Vince P <ve3elb at yahoo.com>
To: APP RPT Forums <app_rpt-users at ohnosec.org> 
Sent: Wednesday, September 4, 2013 11:10 PM
Subject: [App_rpt-users] Setting System Time

Hello all. Can anyone guide me on to how to setting up the time on asterisk. The time was set but for some reason a few weeks ago I noticed that its off by 8 hours. Thank you all for your help. 

73, Vince VE3ELB
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From cypresstower at yahoo.com  Fri Sep  6 03:51:11 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Thu, 5 Sep 2013 20:51:11 -0700 (PDT)
Subject: [App_rpt-users] permit=  echolink.conf
In-Reply-To: <e8569.64b88359.3f5a7f40@aol.com>
References: <e8569.64b88359.3f5a7f40@aol.com>
Message-ID: <1378439471.3337.YahooMailNeo@web163605.mail.gq1.yahoo.com>

Good call on echolink permit list increase by changing the 
EL_MAX_CALL_LIST? 30 to a larger number.? Did you have to recompile.?? 
Would you be kind enough to fill us in on the details??? Thank you

________________________________
From: "N7LD at aol.com" <N7LD at aol.com>
To: tim.sawyer at me.com; app_rpt-users at ohnosec.org 
Sent: Thursday, September 5, 2013 8:43 PM
Subject: Re: [App_rpt-users] permit= echolink.conf



Thanks Tim, that worked!!!
Set mine to 60 and it works.
Lee


In a message dated 9/5/2013 4:58:21 P.M. Pacific Daylight Time, tim.sawyer at me.com writes:
Look at /usr/src/astsrc/asterisk/channels/chan_echolink.c
>
>
>I'm no good at C but I have a stinking suspicion that increasing?EL_MAX_CALL_LIST would allow both bigger permit and deny lists :-)
>--
>Tim
>:wq
>
>On Sep 5, 2013, at 11:25 AM, N7LD at aol.com wrote:
>
>Looks like I am having the same problem. Looks like the MAX stations is 30
>>Lee
>>
>>
>>In a message dated 5/21/2013 11:08:47 A.M. Pacific Daylight Time, w8jtw at yahoo.com writes:
>>Is there anyway to increase the permit=? list in echolink.conf to work with more than 25 or so stations?? Once I get beyond 25 it does not want to allow beyond the first 25.? Or is there a way to limit to just USA connections? 
>>>? ? I dont want to run a wide open echolink but have many friends i don't want to block? ? need to have capacity of about 100 or so.
>>>
>>>Thought of trying Permit=W*,N*,A*,K* not sure if that would limit to just US callsigns or not.
>>>
>>>Any suggestions?
>>>
>>>Thanks
>>>
>>>Joe 
>>>W8JTW
>>>Node 27891? 
>>>_______________________________________________
>>>App_rpt-users mailing list
>>>App_rpt-users at ohnosec.org
>>>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>>>_______________________________________________
>>App_rpt-users mailing list
>>App_rpt-users at ohnosec.org
>>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>>
>
>
>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
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From N7LD at aol.com  Fri Sep  6 04:53:53 2013
From: N7LD at aol.com (N7LD at aol.com)
Date: Fri, 6 Sep 2013 00:53:53 -0400 (EDT)
Subject: [App_rpt-users] permit=  echolink.conf
Message-ID: <2b55a.2ad77636.3f5ab9e0@aol.com>

Yes, you need to recompile after the change.
make upgrade, make svsrc  for Limey
 
 
 
In a message dated 9/5/2013 8:51:12 P.M. Pacific Daylight Time,  
cypresstower at yahoo.com writes:

 
Good call on  echolink permit list increase by changing the 
EL_MAX_CALL_LIST  30 to  a larger number.  Did you have to  recompile.    
Would you be kind enough to fill  us in on the details??? Thank you


 
 
 

From: "N7LD at aol.com"  <N7LD at aol.com>
To:  tim.sawyer at me.com; app_rpt-users at ohnosec.org 
Sent: Thursday, September 5, 2013 8:43  PM
Subject: Re:  [App_rpt-users] permit= echolink.conf



 
Thanks Tim, that worked!!!
Set mine to 60 and it works.
Lee
 
 
 
In a message dated 9/5/2013 4:58:21 P.M. Pacific Daylight Time,  
tim.sawyer at me.com writes:

Look at /usr/src/astsrc/asterisk/channels/chan_echolink.c


I'm no good at C but I have a stinking suspicion that  increasing 
EL_MAX_CALL_LIST would allow both bigger permit and deny  lists :-)
 
--
Tim
:wq



On Sep 5, 2013, at 11:25 AM, _N7LD at aol.com_ (mailto:N7LD at aol.com)  wrote:



Looks like I am having the same problem. Looks like the MAX stations  is 30
Lee
 
 
 
In a message dated 5/21/2013 11:08:47 A.M. Pacific Daylight Time, 
_w8jtw at yahoo.com_ (mailto:w8jtw at yahoo.com)   writes:

Is there anyway  to increase the permit=  list in echolink.conf to work 
with more  than 25 or so stations?  Once I get beyond 25 it does not want to  
allow beyond the first 25.  Or is there a way to limit to just USA  
connections? 
I dont want to run a wide open echolink  but have many friends i don't want 
to block    need to have  capacity of about 100 or so.

Thought of trying Permit=W*,N*,A*,K*  not sure if that would limit to just 
US callsigns or not.

Any  suggestions?

Thanks

Joe 
W8JTW
Node 27891   
_______________________________________________
App_rpt-users  mailing list
_App_rpt-users at ohnosec.org_ (mailto:App_rpt-users at ohnosec.org) 
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



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From telesistant at hotmail.com  Sat Sep  7 01:34:40 2013
From: telesistant at hotmail.com (Jim Duuuude)
Date: Fri, 6 Sep 2013 18:34:40 -0700
Subject: [App_rpt-users] running out of 2XXXX node numbers
Message-ID: <BLU180-W32957068004111D28E0548B13D0@phx.gbl>

Since we were at the 'last' of the 2XXXX 5 digit node numbers, I "reclaimed" some
un-issued node numbers in the 27000-29999 range. We now have 250 node numbers
in the "free" pool that will be issued before going to a new number sequence.

IF any of you have "reservations" for node numbers perhaps "next to" yours that
haven't been assigned yet, speak now or forever hold your peace (or piece, perhaps,
if you are a gun enthusiast :-) ), before what you were wanting/expecting gets given
to someone else.

Thanks

Jim WB6NIL

 		 	   		  
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From randy at neals.ca  Sat Sep  7 08:34:10 2013
From: randy at neals.ca (Randy Neals)
Date: Sat, 7 Sep 2013 03:34:10 -0500
Subject: [App_rpt-users] PCGM (Programmable Clock Generator Module)
Message-ID: <CAHDTD7rm7DFA2e2S-rFVcDQziJukgOC5GuiZLEn+HNah_2JXFQ@mail.gmail.com>

Does anyone have any info on possibility availability of the PCGM?

thx,
-R
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From robert at n5qm.com  Sat Sep  7 15:40:36 2013
From: robert at n5qm.com (Robert Garcia)
Date: Sat, 7 Sep 2013 10:40:36 -0500
Subject: [App_rpt-users] Beagle board Black
In-Reply-To: <2e4001ceaa12$677ea790$367bf6b0$@o2.co.uk>
References: <2e4001ceaa12$677ea790$367bf6b0$@o2.co.uk>
Message-ID: <CAMLJ17OmbpX-y=pL6ug4T_D=_+8FQqiFHzDFGMY1eOfei-DKnA@mail.gmail.com>

Ken,

The BBB requires a 3.2 kernel which has changes that require a rewrite
of some portions of the code from my understanding, so it is currently
not supported.  I expect multiple people are working on it, but to my
knowledge nobody has publicly confirmed a working system.

Robert
N5QM

On Thu, Sep 5, 2013 at 3:32 AM, Ken Robinson <ken-robinson at o2.co.uk> wrote:
>
>
> Hi All,
>
>
>
> Can anyone tell me if the Beagle board Black will act as an Allstar node,
> reading about it I think some of the input outputs have been
>
> taken out of the new board
>
>
>
> 73?s
>
> Ken
>
> G0lce
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>


From kb2ear at kb2ear.net  Sat Sep  7 16:17:25 2013
From: kb2ear at kb2ear.net (Scott Weis)
Date: Sat, 7 Sep 2013 12:17:25 -0400
Subject: [App_rpt-users] Presentation
Message-ID: <3BE8A598-9F1B-49B8-A4C2-9D87F52E365A@kb2ear.net>

Does anyone have a presentation that covers asterisk/app_rpt/allstarlink that they would be willing to share? I have a club presentation on Wednesday and have been totally slammed at work.

Thanks & 73 de,
Scott KB2EAR 

From dshaw at ke6upi.com  Sat Sep  7 20:20:05 2013
From: dshaw at ke6upi.com (David KE6UPI)
Date: Sat, 7 Sep 2013 13:20:05 -0700
Subject: [App_rpt-users] SIP VoIP for Asterisk
In-Reply-To: <ut8ketm5irym5r4lf0kx7fw1.1378420344915@email.android.com>
References: <ut8ketm5irym5r4lf0kx7fw1.1378420344915@email.android.com>
Message-ID: <CAH4RvbLOEK4JPCONZ1EKxMazdZ-v2pOPcLErRwKV0azHfckvrA@mail.gmail.com>

I'm lost.

I was going to reply with a long email asking questions and helping out..
I'm not going too.

Good Luck, David




On Thu, Sep 5, 2013 at 3:32 PM, Dwaine Garden VE3GIF <
DwaineGarden at rogers.com> wrote:

> Bill, sorry for hijacking your thread.  Didn't mean it....
>
> Dwine
>
> Bill South <wbs099 at yahoo.com> wrote:
>
> >     While this thread is not part of what I originally asked about (SIP
> provider recommendations) I'll add that 911 access is mandated in some
> LATAs, but seems to be some debate if it must be 911 or if 9-911 is OK, or
> other variations.  I worked in the telecom business for over 25 years in
> various parts of the USA and different localities would respond differently
> to the question about 911.  When E911 first came out, where a database
> lookup was done by the local police emeregency agency systems answering 911
> calls showing names and addresses, some jurisdictions mandated that
> database had to show the exact location the call was originating from, not
> a billing address of some companies headquarters.  Big problem for some
> large companies where a single billing address is used for all telco
> circuits.  For residences I'm not sure anyone has put much thought into
> laws regarding 911 dialing and any restrictions thereof; businesses on the
> other hand, where
> > 100's maybe thousands of workers are in the same building and floor and
> locating a 911 caller could be pretty tough for local emnergency
> responders, there are, or were anyway, laws in some locations mandating 911
> unrestricted.  The thought has always been though that in an emergency
> people have the expectation, whether at home or work, that they can dial
> 911 and get help; not sure where the laws stands now on that across the USA.
> >
> >--------------------------------------------
> >On Thu, 9/5/13, Dwaine Garden VE3GIF <DwaineGarden at rogers.com> wrote:
> >
> > Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
> > To: "Jim Duuuude" <telesistant at hotmail.com>
> > Cc: "app_rpt mailing list" <app_rpt-users at ohnosec.org>
> > Date: Thursday, September 5, 2013, 7:40 PM
> >
> > Canadian customers have to provide an
> > address and name for 911 purposes for a DID.  Bell and
> > Rogers were mandated with their VoIP services.
> >
> > I gave them the logs of the server where the IP address were
> > listed.   Rogers was good and moved me to a new IP address
> > and filtered out the source.
> >
> > I had an iptables script that would ban IP address that
> > attempted a certain number of connections during a certain
> > amount of time.  Then ban that IP address.  It worked great.
> >  It stopped the hackers script dead cold.  Until they ran
> > the 911 script.  They were ruthless in their attacks.
> >
> > My all-star node was the best fun I have had with Ham radio.
> >  I'll fire it back up again.   Just waiting for the dust
> > to settle without a DID setup.
> >
> > Dwaine
> >
> > Jim Duuuude <telesistant at hotmail.com> wrote:
> >
> > Double BRAVO-SIERRA!!
> >
> > If that were true, a LOT of large business owners (some of
> > which may even operate
> > large businesses :-) ) would be in jail, if they HAPPEN to
> > have a phone system or even
> > worse, service from the "phone company", that
> > requires dialing 9 to get an "outside line".
> >
> > I bet even the police dept has to dial 9 to dial 911.
> >
> > And as far as that goes, put the line in your pet
> > tarantula's name, and make it clear to them
> > that the spider is more then willing to 'serve its
> > time' for such a terrible transgressions!! :-)
> >
> > Wholesale outbound telecom services, such as ones provided
> > by most SIP providers, are *NOT*
> > "in your name", *NOR* do they even technically
> > have a "phone number" or a "service
> > address".
> > Just because you pay the bill for them does not, in any
> > manner, construe that you are the end user
> > of the service.
> >
> > Jim
> >
> >
> >
> >
> > Date: Thu, 5 Sep
> > 2013 14:58:42 -0400
> > Subject: RE: [App_rpt-users] SIP VoIP for Asterisk
> > From: DwaineGarden at rogers.com
> > To: telesistant at hotmail.com
> > CC: dshaw at ke6upi.com; app_rpt-users at ohnosec.org
> >
> > It was an interesting experience.  Has anyone else
> > experienced the same problem and had the police involved?
> > I did mention that I would change the
> > Dial string.  The police told me that a person has to be
> > able to dial 911 and get emergency services.
> >
> > I gave up and just walked over to the all-star computer and
> > hit the power button.  Sadly, it has been off ever since.
> >
> > If I had changed the 911 Dial string before the incident.
> > The police would of not gotten involved. Never known about
> > the situation.
> >
> > Just let everyone know.  The two police officers were very
> > good about it.  They told me that its happening a lot and
> > 911 is getting a little upset about it.
> >
> > I told them they should go after the person responsible.
> > They told me that's you.   The line is in your name.
> >
> > Anyway, just wanted to throw out there my experience.
> >
> > Dwaine
> >
> > Jim Duuuude <telesistant at hotmail.com> wrote:
> >
> > Okay (and yes, that is STUPID and most likely
> > WRONG, but most police depts are
> > just completely ignorant of telecom issues). So, fine...
> > GIVE them access to 911.
> > Let them dial it. But, sadly, on YOUR phone network, the
> > dialing string is just a LITTLE
> > bit longer (like about 30 digits in front of the 911)... get
> > it?
> >
> > Jim
> >
> >
> > Date: Thu, 5 Sep
> > 2013 14:07:58 -0400
> > From: DwaineGarden at rogers.com
> > To: dshaw at ke6upi.com
> > CC: app_rpt-users at ohnosec.org
> > Subject: Re: [App_rpt-users] SIP VoIP for Asterisk
> >
> > The problem is you are not allowed by law to have a phone
> > without unrestricted access to 911.  I had Metro Toronto
> > police at my door explaining that even if I block 911 to any
> > outside connections I would be breaking the law.  If you
> > have a server on the internet with sip.  They have to able
> > to connect to be able to call 911.
> >
> > I told the police it was retard.  They told me that was fine
> > they will charge me.
> >
> > Police told me that even if someone breaks into your house.
> > If there is a phone install,  the criminals better have
> > access to dial 911 unrestrictive.
> >
> > The hackers did not get into the box.  They were trying for
> > months.   Got pissed off and changed their script to dial
> > 911 constantly.  SIP and DID see a 911 call.  It dials it.
> > No questions asked.  No login or nothing.
> >
> > The Police told me it was a huge problem.  SIP or DID are
> > setup like a public pay phone.  Full access to 911.
> >
> > It was an eye opener for me.   You learn something new
> > everyday.  If I see someone asking about SIP or DID.  I let
> > them know about my experience.
> >
> > David KE6UPI <dshaw at ke6upi.com> wrote:
> >
> > I'm sorry Dwaine what are you
> > talking about? Sorry If I don't understand what your
> > talking about.
> >
> > I have both a public Asterisk server and a local
> > Asterisk server. I have never had anyone connect and make a
> > call that was not authenticated user.. Yes they try and
> > fail2ban will block them. There are many way to stop
> > unwanted hackers on your server.
> >
> >
> >
> > As for dialing 911 just make a dial plain to route to space
> > if you want.
> >
> > Google "Asterisk Security"
> >
> > http://www.voip-info.org/wiki/view/Asterisk+security
> >
> >
> >
> > David
> >
> >
> >
> > On Thu, Sep 5, 2013
> > at 9:13 AM, Dwaine Garden VE3GIF <DwaineGarden at rogers.com>
> > wrote:
> >
> > It
> > works great until the hacks find the machine.  They
> > port scan non stop.  Its especially fun when their
> > scripts dial 911 constantly.  There is no way to turn
> > off dialing 911 for SIP.
> >
> >
> >
> >
> > Bill South <wbs099 at yahoo.com>
> > wrote:
> >
> >
> >
> > >     I'm thinking of adding some type of
> > SIP trunking or other VoIP service provider to my ACID
> > Asterisk system to support in/out bound calling.
> >  I've read some emails on the app_rpt reflector
> > with names of providers mentioned, but I am looking for
> > recommendations, as there are zillions of VoIP providers out
> > there.  This is going to be used very sparingly so
> > least-cost is a good thing, but good reliability and no
> > bombardment with email adds by the provider is desired too.
> >  I can easily get by with a single number, but may want
> > to add additional DIDs later.  Thoughts?
> >
> >
> > >
> >
> > >Bill
> >
> > >
> >
> > >
> >
> > >_______________________________________________
> >
> > >App_rpt-users mailing list
> >
> > >App_rpt-users at ohnosec.org
> >
> > >http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> >
> > _______________________________________________
> >
> > App_rpt-users mailing list
> >
> > App_rpt-users at ohnosec.org
> >
> > http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> >
> >
> >
> >
> > _______________________________________________
> > App_rpt-users mailing list
> > App_rpt-users at ohnosec.org
> > http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> >
> > -----Inline Attachment Follows-----
> >
> > _______________________________________________
> > App_rpt-users mailing list
> > App_rpt-users at ohnosec.org
> > http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> >
> >_______________________________________________
> >App_rpt-users mailing list
> >App_rpt-users at ohnosec.org
> >http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
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From hromano at earthlink.net  Sun Sep  8 06:19:08 2013
From: hromano at earthlink.net (Harry Romano)
Date: Sun, 8 Sep 2013 02:19:08 -0400
Subject: [App_rpt-users] announcements
Message-ID: <002101ceac5b$5c8710c0$15953240$@net>

Hi all ,

How do you turn off the announcements in allstar? Is there a way to turn
their volume down?

 

73's

 

Harry  KC4RPP

 

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From brett.friermood at gmail.com  Sun Sep  8 15:52:09 2013
From: brett.friermood at gmail.com (Brett Friermood)
Date: Sun, 8 Sep 2013 10:52:09 -0500
Subject: [App_rpt-users] announcements
In-Reply-To: <002101ceac5b$5c8710c0$15953240$@net>
References: <002101ceac5b$5c8710c0$15953240$@net>
Message-ID: <CADS0AtTy9ofkPFpbtPFZq8Lr_DA0uzDA-sx6D6YLTwdPGQy_-A@mail.gmail.com>

> How do you turn off the announcements in allstar?

http://ohnosec.org/drupal/node/102

>Is there a way to turn their volume down?

http://allstarnode.com/viewtopic.php?f=6&t=26

Brett KQ9N
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From harvard5362 at yahoo.com  Sun Sep  8 17:03:29 2013
From: harvard5362 at yahoo.com (C B)
Date: Sun, 8 Sep 2013 10:03:29 -0700 (PDT)
Subject: [App_rpt-users] Passing DTMF to another controller
Message-ID: <1378659809.43896.YahooMailNeo@web124503.mail.ne1.yahoo.com>

Hi
?
I am trying to get DTMF from other?ALLSTAR nodes to pass to a controller connected to a node.
My configuration is a beagle board with the controller on port 2 (another controller will ne on port 1 latter).
?
I have set in the RPT.CONF 
?
linktolink=yes
propagate_dtmf=yes
?
On both nodes.
?
Below I have copied my RPT.CONF information in case I have placed them in the wrong place or I have a syntax error or some other error.
?
Thank you in advance for any help.
?
?
Chris
?
?
; Radio Repeater configuration file (for use with app_rpt)
;

;
; Your Repeater
;

[29876]???????????????????????????????? ; Change this to your assigned node number
rxchannel=Beagle/beagle29876
duplex=0
erxgain=-3
etxgain=3
;controlstates=controlstates
scheduler=schedule29876
morse=morse29876
macro=macro29876
functions=functions29876
phone_functions=functions29876
link_functions=functions29876
telemetry=telemetry
wait_times=wait-times
context =? radio
callerid = "Repeater" <0000000000>
idrecording = |iN6LXX
accountcode=RADIO
hangtime=001
althangtime=100
totime=170000
idtime=540000
politeid=3000
linktolink=yes
propagate_dtmf=yes
idtalkover=|iN6LXX
unlinkedct=ct2
remotect=ct3
linkunkeyct=ct8
;nolocallinkct=0
;eannmode=1
;connpgm=yourconnectprogram
;discpgm=yourdisconnectprogram
;lnkactenable=0
;lnkacttime=1800
;lnkactmacro=*52
;lnkacttimerwarn=30seconds
;remote_inact_timeout=1800
;remote_timeout=3600
nounkeyct=1
;holdofftelem=0
beaconing=0
;
; *** Status Reporting ***
;
; Uncomment the either group following two statpost lines to report the status of your node to stats.allstarlink.org
; depending on whether you are running ACID or Limey Linux.
; ** For ACID ***
statpost_program=/usr/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
; ** For Limey Linux **
;statpost_program=/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
;statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
;
; Morse code parameters, these are common to all repeaters.
;
[morse29876]
speed=18
frequency=800
amplitude=4096
idfrequency=750
idamplitude=1024
[schedule29876]???????????????????????????????????????????????????????????????? 
;dtmf_function =? m h dom mon dow? ; ala cron, star is implied????????????????? 

[functions29876]
1=ilink,1
2=ilink,2
3=ilink,3
4=ilink,4
5=macro,1
70=ilink,5
71=ilink,6
72=ilink,7
73=ilink,15
74=ilink,16
75=ilink,8
80=status,1
81=status,2
6=autopatchup,noct=1,farenddisconnect=1,dialtime=20000? ; Autopatch up
0=autopatchdn?????????????????????????? ; Autopatch down
989=cop,4
980=status,3
99=cop,6
; Place command macros here
[macro29876]
?

[29877]???????????????????????????????? ; Change this to your assigned node number
rxchannel=Beagle/beagle29877
duplex=0
erxgain=-3
etxgain=3
;controlstates=controlstates
scheduler=schedule29877
morse=morse29877
macro=macro29877
functions=functions29877
phone_functions=functions29877
link_functions=functions29877
telemetry=telemetry
wait_times=wait-times
context =? radio
callerid = "Repeater" <0000000000>
idrecording = |iN6LXX
accountcode=RADIO
hangtime=100
althangtime=100
totime=170000
idtime=540000
politeid=3000
linktolink=yes
propagate_dtmf=yes
idtalkover=|iN6LXX
unlinkedct=ct2
remotect=ct3
linkunkeyct=ct8
;nolocallinkct=0
;eannmode=1
;connpgm=yourconnectprogram
;discpgm=yourdisconnectprogram
;lnkactenable=0
;lnkacttime=1800
;lnkactmacro=*52
;lnkacttimerwarn=30seconds
;remote_inact_timeout=1800
;remote_timeout=3600
nounkeyct=0
;holdofftelem=0
beaconing=0
;
; *** Status Reporting ***
;
; Uncomment the either group following two statpost lines to report the status of your node to stats.allstarlink.org
; depending on whether you are running ACID or Limey Linux.
; ** For ACID ***
statpost_program=/usr/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
; ** For Limey Linux **
;statpost_program=/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
;statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
;
; Morse code parameters, these are common to all repeaters.
;
[morse29877]
speed=18
frequency=800
amplitude=4096
idfrequency=750
idamplitude=1024
[schedule29877]???????????????????????????????????????????????????????????????? 
;dtmf_function =? m h dom mon dow? ; ala cron, star is implied????????????????? 

[functions29877]
1=ilink,1
2=ilink,2
3=ilink,3
4=ilink,4
5=macro,1
70=ilink,5
71=ilink,6
72=ilink,7
73=ilink,15
74=ilink,16
75=ilink,8
80=status,1
81=status,2
6=autopatchup,noct=1,farenddisconnect=1,dialtime=20000? ; Autopatch up
0=autopatchdn?????????????????????????? ; Autopatch down
989=cop,4
980=status,3
99=cop,6
; Place command macros here
[macro29877]
?

[telemetry]
ct1=|t(350,0,100,2048)(500,0,100,2048)(660,0,100,2048)
ct2=|t(660,880,150,2048)
ct3=|t(440,0,150,4096)
ct4=|t(550,0,150,2048)
ct5=|t(660,0,150,2048)
ct6=|t(880,0,150,2048)
ct7=|t(660,440,150,2048)
ct8=|t(700,1100,150,2048)
remotetx=|t(1633,0,50,3000)(0,0,80,0)(1209,0,50,3000);
remotemon=|t(1209,0,50,2048)
cmdmode=|t(900,903,200,2048)
functcomplete=|t(1000,0,100,2048)(0,0,100,0)(1000,0,100,2048)
patchup=rpt/callproceeding
patchdown=rpt/callterminated
;
; This section allows wait times for telemetry events to be adjusted
; A section for wait times can be defined for every repeater
;
[wait-times]??????????????????????????????????????????????????????????????????? 
telemwait=2000
idwait=500
unkeywait=2000
calltermwait=2000
;
; This is where you define your nodes which cam be connected to.
;
[nodes]
; Note, if you are using automatic update for allstar link nodes,
; no allstar link nodes should be defined here. Only place a definition
; for your locak nodes, and private (off of allstar link) nodes here.
29876 = radio at 127.0.0.1/29876,NONE
29877 = radio at 127.0.0.1/29877,NONE
; Memories for remote bases
[memory]
;00 = 146.580,100.0,m
;01 = 147.030,103.5,m+t
;02 = 147.240,103.5,m+t
;03 = 147.765,79.7,m-t
;04 = 146.460,100.0,m
;05 = 146.550,100.0,m

#includeifexists custom/rpt.conf
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From tim.sawyer at me.com  Sun Sep  8 17:23:33 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Sun, 08 Sep 2013 10:23:33 -0700
Subject: [App_rpt-users] announcements
In-Reply-To: <002101ceac5b$5c8710c0$15953240$@net>
References: <002101ceac5b$5c8710c0$15953240$@net>
Message-ID: <DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>

When you set the TX and RX levels per instructions the announcements will be the correct level. 

Please (again) see http://ohnosec.org/drupal/node/48 for all the gory details of setup with chan_usbradio. If you use chan_simpleusb the RX procedure is a bit different: Generate (with a service monitor) a full quieting 1kHz tone at 3kHz dev with no CTCSS. At the OS CLI type radio_tune_menu, adjust the reading for 3kHz. With a RTCM the procedure is pretty much the same except you have nice indicator LEDs right on the RTCM.

Transmit is pretty much the same regardless of the interface.  Turn of any TX CTCSS. Press *989 to generate the reference test level tone. Then measuring with your service monitor set the dev to 3 kHz.

--
Tim
:wq

On Sep 7, 2013, at 11:19 PM, Harry Romano <hromano at earthlink.net> wrote:

> Is there a way to turn their volume down?

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From tim.sawyer at me.com  Sun Sep  8 17:42:11 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Sun, 08 Sep 2013 10:42:11 -0700
Subject: [App_rpt-users] Passing DTMF to another controller
In-Reply-To: <1378659809.43896.YahooMailNeo@web124503.mail.ne1.yahoo.com>
References: <1378659809.43896.YahooMailNeo@web124503.mail.ne1.yahoo.com>
Message-ID: <FE7B5F24-69CE-4972-BA1D-17DFAE05050F@me.com>

This works by connecting to the node that has propagate_dtmf=yes (node 29876 in your case). Then do *40 to enter the command mode. From then on any touch tones (except * and #) will pas out the TX. Then press # to exit command mode and then (*10 to) disconnect. 
--
Tim
:wq

On Sep 8, 2013, at 10:03 AM, C B <harvard5362 at yahoo.com> wrote:

> Hi
>  
> I am trying to get DTMF from other ALLSTAR nodes to pass to a controller connected to a node.
> My configuration is a beagle board with the controller on port 2 (another controller will ne on port 1 latter).
>  
> I have set in the RPT.CONF
>  
> linktolink=yes
> propagate_dtmf=yes
>  
> On both nodes.
>  
> Below I have copied my RPT.CONF information in case I have placed them in the wrong place or I have a syntax error or some other error.
>  
> Thank you in advance for any help.
>  
>  
> Chris
>  
>  
> ; Radio Repeater configuration file (for use with app_rpt)
> ;
> 
> ;
> ; Your Repeater
> ;
> 
> [29876]                                 ; Change this to your assigned node number
> rxchannel=Beagle/beagle29876
> duplex=0
> erxgain=-3
> etxgain=3
> ;controlstates=controlstates
> scheduler=schedule29876
> morse=morse29876
> macro=macro29876
> functions=functions29876
> phone_functions=functions29876
> link_functions=functions29876
> telemetry=telemetry
> wait_times=wait-times
> context =  radio
> callerid = "Repeater" <0000000000>
> idrecording = |iN6LXX
> accountcode=RADIO
> hangtime=001
> althangtime=100
> totime=170000
> idtime=540000
> politeid=3000
> linktolink=yes
> propagate_dtmf=yes
> idtalkover=|iN6LXX
> unlinkedct=ct2
> remotect=ct3
> linkunkeyct=ct8
> ;nolocallinkct=0
> ;eannmode=1
> ;connpgm=yourconnectprogram
> ;discpgm=yourdisconnectprogram
> ;lnkactenable=0
> ;lnkacttime=1800
> ;lnkactmacro=*52
> ;lnkacttimerwarn=30seconds
> ;remote_inact_timeout=1800
> ;remote_timeout=3600
> nounkeyct=1
> ;holdofftelem=0
> beaconing=0
> ;
> ; *** Status Reporting ***
> ;
> ; Uncomment the either group following two statpost lines to report the status of your node to stats.allstarlink.org
> ; depending on whether you are running ACID or Limey Linux.
> ; ** For ACID ***
> statpost_program=/usr/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
> statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
> ; ** For Limey Linux **
> ;statpost_program=/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
> ;statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
> ;
> ; Morse code parameters, these are common to all repeaters.
> ;
> [morse29876]
> speed=18
> frequency=800
> amplitude=4096
> idfrequency=750
> idamplitude=1024
> [schedule29876]                                                                 
> ;dtmf_function =  m h dom mon dow  ; ala cron, star is implied                 
> 
> [functions29876]
> 1=ilink,1
> 2=ilink,2
> 3=ilink,3
> 4=ilink,4
> 5=macro,1
> 70=ilink,5
> 71=ilink,6
> 72=ilink,7
> 73=ilink,15
> 74=ilink,16
> 75=ilink,8
> 80=status,1
> 81=status,2
> 6=autopatchup,noct=1,farenddisconnect=1,dialtime=20000  ; Autopatch up
> 0=autopatchdn                           ; Autopatch down
> 989=cop,4
> 980=status,3
> 99=cop,6
> ; Place command macros here
> [macro29876]
>  
> 
> [29877]                                 ; Change this to your assigned node number
> rxchannel=Beagle/beagle29877
> duplex=0
> erxgain=-3
> etxgain=3
> ;controlstates=controlstates
> scheduler=schedule29877
> morse=morse29877
> macro=macro29877
> functions=functions29877
> phone_functions=functions29877
> link_functions=functions29877
> telemetry=telemetry
> wait_times=wait-times
> context =  radio
> callerid = "Repeater" <0000000000>
> idrecording = |iN6LXX
> accountcode=RADIO
> hangtime=100
> althangtime=100
> totime=170000
> idtime=540000
> politeid=3000
> linktolink=yes
> propagate_dtmf=yes
> idtalkover=|iN6LXX
> unlinkedct=ct2
> remotect=ct3
> linkunkeyct=ct8
> ;nolocallinkct=0
> ;eannmode=1
> ;connpgm=yourconnectprogram
> ;discpgm=yourdisconnectprogram
> ;lnkactenable=0
> ;lnkacttime=1800
> ;lnkactmacro=*52
> ;lnkacttimerwarn=30seconds
> ;remote_inact_timeout=1800
> ;remote_timeout=3600
> nounkeyct=0
> ;holdofftelem=0
> beaconing=0
> ;
> ; *** Status Reporting ***
> ;
> ; Uncomment the either group following two statpost lines to report the status of your node to stats.allstarlink.org
> ; depending on whether you are running ACID or Limey Linux.
> ; ** For ACID ***
> statpost_program=/usr/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
> statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
> ; ** For Limey Linux **
> ;statpost_program=/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
> ;statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
> ;
> ; Morse code parameters, these are common to all repeaters.
> ;
> [morse29877]
> speed=18
> frequency=800
> amplitude=4096
> idfrequency=750
> idamplitude=1024
> [schedule29877]                                                                 
> ;dtmf_function =  m h dom mon dow  ; ala cron, star is implied                 
> 
> [functions29877]
> 1=ilink,1
> 2=ilink,2
> 3=ilink,3
> 4=ilink,4
> 5=macro,1
> 70=ilink,5
> 71=ilink,6
> 72=ilink,7
> 73=ilink,15
> 74=ilink,16
> 75=ilink,8
> 80=status,1
> 81=status,2
> 6=autopatchup,noct=1,farenddisconnect=1,dialtime=20000  ; Autopatch up
> 0=autopatchdn                           ; Autopatch down
> 989=cop,4
> 980=status,3
> 99=cop,6
> ; Place command macros here
> [macro29877]
>  
> 
> [telemetry]
> ct1=|t(350,0,100,2048)(500,0,100,2048)(660,0,100,2048)
> ct2=|t(660,880,150,2048)
> ct3=|t(440,0,150,4096)
> ct4=|t(550,0,150,2048)
> ct5=|t(660,0,150,2048)
> ct6=|t(880,0,150,2048)
> ct7=|t(660,440,150,2048)
> ct8=|t(700,1100,150,2048)
> remotetx=|t(1633,0,50,3000)(0,0,80,0)(1209,0,50,3000);
> remotemon=|t(1209,0,50,2048)
> cmdmode=|t(900,903,200,2048)
> functcomplete=|t(1000,0,100,2048)(0,0,100,0)(1000,0,100,2048)
> patchup=rpt/callproceeding
> patchdown=rpt/callterminated
> ;
> ; This section allows wait times for telemetry events to be adjusted
> ; A section for wait times can be defined for every repeater
> ;
> [wait-times]                                                                    
> telemwait=2000
> idwait=500
> unkeywait=2000
> calltermwait=2000
> ;
> ; This is where you define your nodes which cam be connected to.
> ;
> [nodes]
> ; Note, if you are using automatic update for allstar link nodes,
> ; no allstar link nodes should be defined here. Only place a definition
> ; for your locak nodes, and private (off of allstar link) nodes here.
> 29876 = radio at 127.0.0.1/29876,NONE
> 29877 = radio at 127.0.0.1/29877,NONE
> ; Memories for remote bases
> [memory]
> ;00 = 146.580,100.0,m
> ;01 = 147.030,103.5,m+t
> ;02 = 147.240,103.5,m+t
> ;03 = 147.765,79.7,m-t
> ;04 = 146.460,100.0,m
> ;05 = 146.550,100.0,m
> 
> #includeifexists custom/rpt.conf
> 
>  
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From harvard5362 at yahoo.com  Sun Sep  8 17:51:58 2013
From: harvard5362 at yahoo.com (C B)
Date: Sun, 8 Sep 2013 10:51:58 -0700 (PDT)
Subject: [App_rpt-users] Passing DTMF to another controller
In-Reply-To: <FE7B5F24-69CE-4972-BA1D-17DFAE05050F@me.com>
References: <1378659809.43896.YahooMailNeo@web124503.mail.ne1.yahoo.com>
	<FE7B5F24-69CE-4972-BA1D-17DFAE05050F@me.com>
Message-ID: <1378662718.40850.YahooMailNeo@web124505.mail.ne1.yahoo.com>

Tim
?
Thank you that was the step I was missing!!!!
?
After following your instructions it worked.
?
Chris
 

________________________________
 From: Tim Sawyer <tim.sawyer at me.com>
To: C B <harvard5362 at yahoo.com> 
Cc: "app_rpt-users at ohnosec.org" <app_rpt-users at ohnosec.org> 
Sent: Sunday, September 8, 2013 10:42 AM
Subject: Re: [App_rpt-users] Passing DTMF to another controller
  


This works by connecting to the node that has propagate_dtmf=yes (node 29876 in your case). Then do *40 to enter the command mode. From then on any touch tones (except * and #) will pas out the TX. Then press # to exit command mode and then (*10 to) disconnect.?

--
Tim
:wq  

On Sep 8, 2013, at 10:03 AM, C B <harvard5362 at yahoo.com> wrote:

Hi
>
>I am trying to get DTMF from other?ALLSTAR nodes to pass to a controller connected to a node.
>My configuration is a beagle board with the controller on port 2 (another controller will ne on port 1 latter).
>
>I have set in the RPT.CONF 
>
>linktolink=yes
>propagate_dtmf=yes
>
>On both nodes.
>
>Below I have copied my RPT.CONF information in case I have placed them in the wrong place or I have a syntax error or some other error.
>
>Thank you in advance for any help.
>
>
>Chris
>
>
>; Radio Repeater configuration file (for use with app_rpt)
>;
>
>;
>; Your
 Repeater
>;
>
>[29876]???????????????????????????????? ; Change this to your assigned node number
>rxchannel=Beagle/beagle29876
>duplex=0
>erxgain=-3
>etxgain=3
>;controlstates=controlstates
>scheduler=schedule29876
>morse=morse29876
>macro=macro29876
>functions=functions29876
>phone_functions=functions29876
>link_functions=functions29876
>telemetry=telemetry
>wait_times=wait-times
>context =? radio
>callerid = "Repeater" <0000000000>
>idrecording =
 |iN6LXX
>accountcode=RADIO
>hangtime=001
>althangtime=100
>totime=170000
>idtime=540000
>politeid=3000
>linktolink=yes
>propagate_dtmf=yes
>idtalkover=|iN6LXX
>unlinkedct=ct2
>remotect=ct3
>linkunkeyct=ct8
>;nolocallinkct=0
>;eannmode=1
>;connpgm=yourconnectprogram
>;discpgm=yourdisconnectprogram
>;lnkactenable=0
>;lnkacttime=1800
>;lnkactmacro=*52
>;lnkacttimerwarn=30seconds
>;remote_inact_timeout=1800
>;remote_timeout=3600
>nounkeyct=1
>;holdofftelem=0
>beaconing=0
>;
>; *** Status Reporting ***
>;
>; Uncomment the either group following two statpost lines to report the status of your node to stats.allstarlink.org
>; depending on whether you are running ACID or Limey Linux.
>; ** For ACID ***
>statpost_program=/usr/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
>statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
>; ** For Limey Linux **
>;statpost_program=/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
>;statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
>;
>; Morse code parameters, these are common to all repeaters.
>;
>[morse29876]
>speed=18
>frequency=800
>amplitude=4096
>idfrequency=750
>idamplitude=1024
>[schedule29876]???????????????????????????????????????????????????????????????? 
>;dtmf_function =? m h dom mon dow? ; ala cron, star is implied????????????????? 
>
>[functions29876]
>1=ilink,1
>2=ilink,2
>3=ilink,3
>4=ilink,4
>5=macro,1
>70=ilink,5
>71=ilink,6
>72=ilink,7
>73=ilink,15
>74=ilink,16
>75=ilink,8
>80=status,1
>81=status,2
>6=autopatchup,noct=1,farenddisconnect=1,dialtime=20000? ; Autopatch up
>0=autopatchdn?????????????????????????? ; Autopatch down
>989=cop,4
>980=status,3
>99=cop,6
>; Place command macros here
>[macro29876]
>
>
>[29877]???????????????????????????????? ; Change this to your assigned node
 number
>rxchannel=Beagle/beagle29877
>duplex=0
>erxgain=-3
>etxgain=3
>;controlstates=controlstates
>scheduler=schedule29877
>morse=morse29877
>macro=macro29877
>functions=functions29877
>phone_functions=functions29877
>link_functions=functions29877
>telemetry=telemetry
>wait_times=wait-times
>context =? radio
>callerid = "Repeater" <0000000000>
>idrecording = |iN6LXX
>accountcode=RADIO
>hangtime=100
>althangtime=100
>totime=170000
>idtime=540000
>politeid=3000
>linktolink=yes
>propagate_dtmf=yes
>idtalkover=|iN6LXX
>unlinkedct=ct2
>remotect=ct3
>linkunkeyct=ct8
>;nolocallinkct=0
>;eannmode=1
>;connpgm=yourconnectprogram
>;discpgm=yourdisconnectprogram
>;lnkactenable=0
>;lnkacttime=1800
>;lnkactmacro=*52
>;lnkacttimerwarn=30seconds
>;remote_inact_timeout=1800
>;remote_timeout=3600
>nounkeyct=0
>;holdofftelem=0
>beaconing=0
>;
>; *** Status Reporting
 ***
>;
>; Uncomment the either group following two statpost lines to report the status of your node to stats.allstarlink.org
>; depending on whether you are running ACID or Limey Linux.
>; ** For ACID ***
>statpost_program=/usr/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
>statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
>; ** For Limey Linux **
>;statpost_program=/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null
>;statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates
>;
>; Morse code parameters, these are common to all
 repeaters.
>;
>[morse29877]
>speed=18
>frequency=800
>amplitude=4096
>idfrequency=750
>idamplitude=1024
>[schedule29877]???????????????????????????????????????????????????????????????? 
>;dtmf_function =? m h dom mon dow? ; ala cron, star is implied????????????????? 
>
>[functions29877]
>1=ilink,1
>2=ilink,2
>3=ilink,3
>4=ilink,4
>5=macro,1
>70=ilink,5
>71=ilink,6
>72=ilink,7
>73=ilink,15
>74=ilink,16
>75=ilink,8
>80=status,1
>81=status,2
>6=autopatchup,noct=1,farenddisconnect=1,dialtime=20000? ; Autopatch up
>0=autopatchdn?????????????????????????? ; Autopatch down
>989=cop,4
>980=status,3
>99=cop,6
>; Place command macros here
>[macro29877]
>
>
>[telemetry]
>ct1=|t(350,0,100,2048)(500,0,100,2048)(660,0,100,2048)
>ct2=|t(660,880,150,2048)
>ct3=|t(440,0,150,4096)
>ct4=|t(550,0,150,2048)
>ct5=|t(660,0,150,2048)
>ct6=|t(880,0,150,2048)
>ct7=|t(660,440,150,2048)
>ct8=|t(700,1100,150,2048)
>remotetx=|t(1633,0,50,3000)(0,0,80,0)(1209,0,50,3000);
>remotemon=|t(1209,0,50,2048)
>cmdmode=|t(900,903,200,2048)
>functcomplete=|t(1000,0,100,2048)(0,0,100,0)(1000,0,100,2048)
>patchup=rpt/callproceeding
>patchdown=rpt/callterminated
>;
>; This section allows wait times for telemetry events to be adjusted
>; A section for wait times can be defined for every
 repeater
>;
>[wait-times]??????????????????????????????????????????????????????????????????? 
>telemwait=2000
>idwait=500
>unkeywait=2000
>calltermwait=2000
>;
>; This is where you define your nodes which cam be connected to.
>;
>[nodes]
>; Note, if you are using automatic update for allstar link nodes,
>; no allstar link nodes should be defined here. Only place a definition
>; for your locak nodes, and private (off of allstar link) nodes here.
>29876 = radio at 127.0.0.1/29876,NONE
>29877 = radio at 127.0.0.1/29877,NONE
>; Memories for remote bases
>[memory]
>;00 = 146.580,100.0,m
>;01 = 147.030,103.5,m+t
>;02 = 147.240,103.5,m+t
>;03 = 147.765,79.7,m-t
>;04 = 146.460,100.0,m
>;05 = 146.550,100.0,m
>
>#includeifexists custom/rpt.conf
>
>?_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
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From ke2n at cs.com  Sun Sep  8 18:16:37 2013
From: ke2n at cs.com (Ken)
Date: Sun, 8 Sep 2013 14:16:37 -0400
Subject: [App_rpt-users] announcements
Message-ID: <004601ceacbf$8f38fa10$adaaee30$@cs.com>

>>When you set the TX and RX levels per instructions the announcements will
be the correct level.

Actually Harry, many people find the announcements too loud for their
tastes, even when everything is adjusted OK.

A while back somebody posted a script that went through the entire directory
containing all the announcements and lowered the volume by a certain number
of DB. Perhaps with some searching you could find it, or some other user on
here will have a copy.

73

Ken



From kuggie at kuggie.com  Sun Sep  8 19:40:55 2013
From: kuggie at kuggie.com (Kevin Custer)
Date: Sun, 08 Sep 2013 15:40:55 -0400
Subject: [App_rpt-users] announcements
In-Reply-To: <DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
Message-ID: <522CD2C7.8010002@kuggie.com>

On 9/8/2013 1:23 PM, Tim Sawyer wrote:
> When you set the TX and RX levels per instructions the announcements 
> will be the correct level.

With all due respect, what is a "correct announcement level" level is up 
for debate.  (I'm not bringing this up to change or start a thread on 
why one distro is better than another - and I'd appreciate if no one 
else did either).

When I got into this technology, one of my biggest gripes, along with 
many users, was the level of the telemetry.  Allison just stepped on 
folks, especially if they were a little soft spoken.
Of course, this IS with the levels set properly.

This is one area where XIPAR has an advantage.

In zipper, we can set the Nominal Telemetry Level and the Duck Level.  I 
run -10 dB nominal, and -15 dB duck.  With these levels, Allison doesn't 
blast you out of the water, and if someone (somewhere) has the mic 
keyed, she's ducked out an additional amount, BUT you can still hear 
what she is saying - even along with a loud mouth (such as myself).  
While I refer to "she" (Allison) all telemetry, including CWID's and 
whatever is controlled by these settings.

Thanks,
Kevin




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From buddy at brannan.name  Sun Sep  8 20:38:15 2013
From: buddy at brannan.name (Buddy Brannan)
Date: Sun, 8 Sep 2013 16:38:15 -0400
Subject: [App_rpt-users] announcements
In-Reply-To: <522CD2C7.8010002@kuggie.com>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<522CD2C7.8010002@kuggie.com>
Message-ID: <5C6B91E2-601F-41F2-B15D-6D18BA953FAD@brannan.name>

I agree that this is a great feature that Xipar has. Also subjective, and I don?t know what they?re doing differently, but I have an easier time with level adjustments with Xipar (the automatic level setting for rxnoise seems to work a bit better), and at least on my radios, audio ?sounds? better, better being kind of relative, yes, I understand that. However, it looks like xipar is missing a couple things that main line acide distro has, I mean besides that acid works with Allstar?s autopatch and dial-in service without additional fiddling. Doesn?t appear Xipar has the provision for recording or streaming from app_rpt. Also, it looks like ID pitch and volume are fixed (and a bit lo of my taste), and the echo link channel driver didn?t compile last time I tried a beta of it several months ago. And i at least don?t really need or want Freepbx. So, until or unless Echolink gets worked out along with streaming/recording, I?m sticking with acid, even in spite of some of xipar?s real advantages. 
--
Buddy Brannan, KB5ELV - Erie, PA
Phone: (814) 860-3194 or 888-75-BUDDY



On Sep 8, 2013, at 3:40 PM, Kevin Custer <kuggie at kuggie.com> wrote:

> On 9/8/2013 1:23 PM, Tim Sawyer wrote:
>> When you set the TX and RX levels per instructions the announcements will be the correct level.
> 
> With all due respect, what is a "correct announcement level" level is up for debate.  (I'm not bringing this up to change or start a thread on why one distro is better than another - and I'd appreciate if no one else did either).  
> 
> When I got into this technology, one of my biggest gripes, along with many users, was the level of the telemetry.  Allison just stepped on folks, especially if they were a little soft spoken.
> Of course, this IS with the levels set properly.
> 
> This is one area where XIPAR has an advantage.  
> 
> In zipper, we can set the Nominal Telemetry Level and the Duck Level.  I run -10 dB nominal, and -15 dB duck.  With these levels, Allison doesn't blast you out of the water, and if someone (somewhere) has the mic keyed, she's ducked out an additional amount, BUT you can still hear what she is saying - even along with a loud mouth (such as myself).  While I refer to "she" (Allison) all telemetry, including CWID's and whatever is controlled by these settings. 
> 
> Thanks,
> Kevin
> 
> 
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



From bdboyle at bdboyle.com  Mon Sep  9 00:16:00 2013
From: bdboyle at bdboyle.com (Bryan D. Boyle)
Date: Sun, 08 Sep 2013 20:16:00 -0400
Subject: [App_rpt-users] announcements
In-Reply-To: <5C6B91E2-601F-41F2-B15D-6D18BA953FAD@brannan.name>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<522CD2C7.8010002@kuggie.com>
	<5C6B91E2-601F-41F2-B15D-6D18BA953FAD@brannan.name>
Message-ID: <522D1340.7020000@bdboyle.com>

On 9/8/2013 4:38 PM, Buddy Brannan wrote:
> I agree that this is a great feature that Xipar has. Also subjective, and I don?t know what they?re doing differently, but I have an easier time with level adjustments with Xipar (the automatic level setting for rxnoise seems to work a bit better), and at least on my radios, audio ?sounds? better, better being kind of relative, yes, I understand that. However, it looks like xipar is missing a couple things that main line acide distro has, I mean besides that acid works with Allstar?s autopatch and dial-in service without additional fiddling. Doesn?t appear Xipar has the provision for recording or streaming from app_rpt. Also, it looks like ID pitch and volume are fixed (and a bit lo of my taste), and the echo link channel driver didn?t compile last time I tried a beta of it several months ago. And i at least don?t really need or want Freepbx. So, until or unless Echolink gets worked out along with streaming/recording, I?m sticking with acid, even in spite of some of xipar?s
  real a
dvantages. 
> --

You know, what's interesting, is the different focus of the two and how
they've developed on parallel, but, in some respects, divergent paths:
open source is like that.

I think a couple things, from my standpoint, that I'd like to see in
acid would be including DCS as an option for coded squelch.  IIRC, XIPAR
has it.  That, and the variable audio level features Kevin described.
Other than that?  Both are viable, excellent programs that address a
need...you pays your money, you takes your chances.

So, they're both good; guess it just comes down to what you need to do,
how you addressed it with what tools you had at the time, and how much
pain it would be to migrate (though, to tell the truth, I'd like to try
WN3A's rewrite of one of the codecs...I hear the quality is superb...)

Bryan




From kuggie at kuggie.com  Mon Sep  9 00:21:54 2013
From: kuggie at kuggie.com (Kevin Custer)
Date: Sun, 08 Sep 2013 20:21:54 -0400
Subject: [App_rpt-users] announcements
In-Reply-To: <5C6B91E2-601F-41F2-B15D-6D18BA953FAD@brannan.name>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<522CD2C7.8010002@kuggie.com>
	<5C6B91E2-601F-41F2-B15D-6D18BA953FAD@brannan.name>
Message-ID: <522D14A2.8030508@kuggie.com>

On 9/8/2013 4:38 PM, Buddy Brannan wrote:
> I agree that this is a great feature that Xipar has. Also subjective, and I don't know what they're doing differently, but I have an easier time with level adjustments with Xipar (the automatic level setting for rxnoise seems to work a bit better), and at least on my radios, audio "sounds" better, better being kind of relative, yes, I understand that.

I guess it depends on if you are using the audio w-i-d-e-n-i-n-g filter 
rules.  If you are - then they are not relative, but real improvements 
in audio quality.
In zipper, the following apply:

rxlpf=x
txlpf=x
txhpf=x
txlpf=x

x can be 0, 1 or 2.
More info here:
http://www.wanrepeater.com/xipar_urd_filters.doc




>   However, it looks like xipar is missing a couple things that main line acide distro has, I mean besides that acid works with Allstar's autopatch and dial-in service without additional fiddling. Doesn't appear Xipar has the provision for recording or streaming from app_rpt.

Recording and Streaming - sure it does.  We record everything and it is 
streamed on Radio Reference.
Contact Brian Burton - KB3ORS for details on this.


>   Also, it looks like ID pitch and volume are fixed (and a bit lo of my taste),

You have to put in the following in rpt.conf then you can set it however 
you like.
morse = morse
Then, adjust the settings in the [morse] stanza.


>   and the echo link channel driver didn't compile last time I tried a beta of it several months ago.

Not sure - I personally don't do Echolink, but it worked the last time 
someone else did a new install and connected to WAN.


>   And i at least don't really need or want Freepbx. So, until or unless Echolink gets worked out along with streaming/recording, I'm sticking with acid, even in spite of some of xipar's real advantages.

That's why there are several distros.

IMHO there are two BIG disadvantages currently with XIPAR - No 
WebTransceiver and no AllMon.   So, I run two servers with ACID - all 
fixed...

Kevin



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From buddy at brannan.name  Mon Sep  9 00:37:41 2013
From: buddy at brannan.name (Buddy Brannan)
Date: Sun, 8 Sep 2013 20:37:41 -0400
Subject: [App_rpt-users] announcements
In-Reply-To: <522D1340.7020000@bdboyle.com>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<522CD2C7.8010002@kuggie.com>
	<5C6B91E2-601F-41F2-B15D-6D18BA953FAD@brannan.name>
	<522D1340.7020000@bdboyle.com>
Message-ID: <A3CAB20E-744F-4861-AFCE-ED2948C574D4@brannan.name>

Rewrite of a codec? Hmm, I?m intrigued. 
--
Buddy Brannan, KB5ELV - Erie, PA
Phone: (814) 860-3194 or 888-75-BUDDY



On Sep 8, 2013, at 8:16 PM, Bryan D. Boyle <bdboyle at bdboyle.com> wrote:

> On 9/8/2013 4:38 PM, Buddy Brannan wrote:
>> I agree that this is a great feature that Xipar has. Also subjective, and I don?t know what they?re doing differently, but I have an easier time with level adjustments with Xipar (the automatic level setting for rxnoise seems to work a bit better), and at least on my radios, audio ?sounds? better, better being kind of relative, yes, I understand that. However, it looks like xipar is missing a couple things that main line acide distro has, I mean besides that acid works with Allstar?s autopatch and dial-in service without additional fiddling. Doesn?t appear Xipar has the provision for recording or streaming from app_rpt. Also, it looks like ID pitch and volume are fixed (and a bit lo of my taste), and the echo link channel driver didn?t compile last time I tried a beta of it several months ago. And i at least don?t really need or want Freepbx. So, until or unless Echolink gets worked out along with streaming/recording, I?m sticking with acid, even in spite of some of xipar?s
>  real a
> dvantages. 
>> --
> 
> You know, what's interesting, is the different focus of the two and how
> they've developed on parallel, but, in some respects, divergent paths:
> open source is like that.
> 
> I think a couple things, from my standpoint, that I'd like to see in
> acid would be including DCS as an option for coded squelch.  IIRC, XIPAR
> has it.  That, and the variable audio level features Kevin described.
> Other than that?  Both are viable, excellent programs that address a
> need...you pays your money, you takes your chances.
> 
> So, they're both good; guess it just comes down to what you need to do,
> how you addressed it with what tools you had at the time, and how much
> pain it would be to migrate (though, to tell the truth, I'd like to try
> WN3A's rewrite of one of the codecs...I hear the quality is superb...)
> 
> Bryan
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



From bdboyle at bdboyle.com  Mon Sep  9 00:47:08 2013
From: bdboyle at bdboyle.com (Bryan D. Boyle)
Date: Sun, 08 Sep 2013 20:47:08 -0400
Subject: [App_rpt-users] announcements
In-Reply-To: <A3CAB20E-744F-4861-AFCE-ED2948C574D4@brannan.name>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<522CD2C7.8010002@kuggie.com>
	<5C6B91E2-601F-41F2-B15D-6D18BA953FAD@brannan.name>
	<522D1340.7020000@bdboyle.com>
	<A3CAB20E-744F-4861-AFCE-ED2948C574D4@brannan.name>
Message-ID: <522D1A8C.4020100@bdboyle.com>

On 9/8/2013 8:37 PM, Buddy Brannan wrote:
> Rewrite of a codec? Hmm, I?m intrigued. 

Should have said rewrite of tx/rx bandpass filtering rules...We talked
about it a little bit while I was working for him a month ago while on
the bench.

My bad, and fingers worked ahead of my brain, listening to 2 sunday
night nets and typing mail messages...sorry...

BB





From kuggie at kuggie.com  Mon Sep  9 02:57:26 2013
From: kuggie at kuggie.com (Kevin Custer)
Date: Sun, 08 Sep 2013 22:57:26 -0400
Subject: [App_rpt-users] announcements
In-Reply-To: <522D1A8C.4020100@bdboyle.com>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<522CD2C7.8010002@kuggie.com>
	<5C6B91E2-601F-41F2-B15D-6D18BA953FAD@brannan.name>
	<522D1340.7020000@bdboyle.com>
	<A3CAB20E-744F-4861-AFCE-ED2948C574D4@brannan.name>
	<522D1A8C.4020100@bdboyle.com>
Message-ID: <522D3916.7030509@kuggie.com>


> On 9/8/2013 8:37 PM, Buddy Brannan wrote:
>> Rewrite of a codec? Hmm, I'm intrigued.
> On 9/8/2013 8:47 PM, Bryan D. Boyle wrote:
>
> Should have said rewrite of tx/rx bandpass filtering rules...We talked
> about it a little bit while I was working for him a month ago while on
> the bench.

Actually - you are correct Bryan.....

When Jeff DePolo was first getting into this technology, he discovered 
problems with the receiver input filter.  Jeff is a perfectionist, and 
like me is an audio purist.

There is a brief reference to his work (detailed below) in the link I 
shared:
http://www.wanrepeater.com/xipar_urd_filters.doc

Here is an email Jeff wrote while testing:

/I spent most of today on a hair-pulling excursion to try to find the cause of some really hellacious aliasing distortion through the repeater.  I could give you the long story some other time, but the short version is that I found a bug in the rx frontend FIR code where the history buffer is shifted to add a new sample.  The line:

memmove(x+1, x, nx-1)

should be

memmove(x+1, x, (nx-1)*2)

Gotta move 16-bit ints not bytes.

With that corrected, the aliasing distortion dropped immensely as you might imagine.  Aliasing-related distortion products are now below -40 dBr as compared to as little as -6 dBr previously!

However, I'm still not happy with the audio.  I'm seeing 1-2% THD through Asterisk from dongle Rx port to Tx port.  That led me down yet another path to quantify the performance of the hardware (DMK Engineering dongle) via my Audio Precision test set, analysis software in Windows (had the dongle connected to an XP machine for testing), etc..  For the most part, it passes, with the basic specs coming fairly close to the specs for the CM108.
It still has other issues, but none that would account for the THD that I'm seeing.  There's also a problem somewhere that manifests as excruciatingly high distortion below about 450 Hz when the input amplitude is over maybe -6
dBFS (I haven't dug into that deeply yet, but I'm guessing it's an overflow/underflow problem in 16 bit integer math somewhere).  That's where I'm stopping for the night.

Anyway, thought you'd want to know about the FIR bug if nothing else.

Please keep in mind that I'm NOT trying to find faults in your code.  I have some perfectionist tendencies, and they really come through when it comes to audio.  If you're open to constructive criticism I'll pass along whatever else I find.  If not, just tell me to go away and I will.

				--- Jeff/

Jeff also wrote a followup:

/I re-did all of the FIR filters, found a few other bugs along the way, etc.  There is still a problem in the preemphasis/deemphasis routines that can result in integer overflows under certain conditions, and I have some other things I want to fix, but I'm going to stop for now until I have some more time to spend on it.//
/
Ultimately, Jeff went through everything and got it to a state where he was satisfied.  I'm fairly sure all of these improvements exist in ACID as well as XIPAR, maybe Dude can verify?

Thanks,
Kevin


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From tim.sawyer at me.com  Mon Sep  9 03:23:42 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Sun, 08 Sep 2013 20:23:42 -0700
Subject: [App_rpt-users] announcements
In-Reply-To: <522CD2C7.8010002@kuggie.com>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<522CD2C7.8010002@kuggie.com>
Message-ID: <DAA8A3C4-0BE9-471B-9CAF-FCBDFC917A13@me.com>

Well we can agree to disagree on Allison's level. I don't find her too loud and none of our group has complained. We do have do have the telemdefault=2 because all the connects and disconnects are annoying at any level.  

Having said that, I wish ACID had audio ducking not so much for Allison but for links on monitor. It's nice to monitor a link (my two meter repeater from UHF for example) and be able to talk over it and have the link audio duck a few dB.
--
Tim
:wq

On Sep 8, 2013, at 12:40 PM, Kevin Custer <kuggie at kuggie.com> wrote:

> In zipper, we can set the Nominal Telemetry Level and the Duck Level.  I run -10 dB nominal, and -15 dB duck.  With these levels, Allison doesn't blast you out of the water, and if someone (somewhere) has the mic keyed, she's ducked out an additional amount, BUT you can still hear what she is saying - even along with a loud mouth (such as myself).  While I refer to "she" (Allison) all telemetry, including CWID's and whatever is controlled by these settings. 

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From tim.sawyer at me.com  Mon Sep  9 05:01:21 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Sun, 08 Sep 2013 22:01:21 -0700
Subject: [App_rpt-users] announcements
In-Reply-To: <000a01cead12$b36e9b30$1a4bd190$@net>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<000a01cead12$b36e9b30$1a4bd190$@net>
Message-ID: <C71C5AA8-5BEC-4D8B-8EC7-81B8DA8BDA74@me.com>

Yes it's easy to turn off dis/connect messages. See telemdefault http://ohnosec.org/drupal/node/103. I like mode 2. 

Echolink levels can be adjusted independently of other audio. In your node stanza you should see the following statements near the top. Uncomment or add them. 

erxgain=-3                          ; Echolink receive gain adjustment
                                    ; Note: Gain is in db-volts (20logVI/VO)
etxgain=3                           ; Echolink transmit gain adjustment
                                    ; Note: Gain is in db-volts (20logVI/VO)

One thing I like to do is add another node to my system just for echolink. That way you can dis/connect it as needed.
--
Tim
:wq

On Sep 8, 2013, at 9:11 PM, Harry Romano <hromano at earthlink.net> wrote:

> Hi Tim,
>  
> Thanks for the reply . I think I want to shut them off. When I was on the win system the other night there were so many connects and disconnect announcements you could not here the net control. Is It easier to just turn them off instead of  turning them down?
>  
> I have the radios adjusted good for input and output deviation, What I am speaking of is the announcements.  Thanks for the info . Apparently there is a difference when you go to radio tune mode from Cli vs. root.  The cli allows more detailed tuning.
>  
> Another issue I have is with Echolink. I personally don?t like Echolink but one of my repeater users does a net with it . The echo link come in at a different volume level than let?s say the web transceiver or another allstar node . both my nodes connect together great but Echolink is defiantly lower . Is there a way to get a volume increase just for Echolink? I was thinking of putting a txmixer = 250 command in the Echolink.conf. But I think that may make everything else get louder too.
>  
>  
> 73's
>  
> Harry  KC4RPP
>  
>  
> From: Tim Sawyer [mailto:tim.sawyer at me.com] 
> Sent: Sunday, September 08, 2013 1:24 PM
> To: Harry Romano
> Cc: app_rpt-users at ohnosec.org
> Subject: Re: [App_rpt-users] announcements
>  
> When you set the TX and RX levels per instructions the announcements will be the correct level. 
>  
> Please (again) see http://ohnosec.org/drupal/node/48 for all the gory details of setup with chan_usbradio. If you use chan_simpleusb the RX procedure is a bit different: Generate (with a service monitor) a full quieting 1kHz tone at 3kHz dev with no CTCSS. At the OS CLI type radio_tune_menu, adjust the reading for 3kHz. With a RTCM the procedure is pretty much the same except you have nice indicator LEDs right on the RTCM.
>  
> Transmit is pretty much the same regardless of the interface.  Turn of any TX CTCSS. Press *989 to generate the reference test level tone. Then measuring with your service monitor set the dev to 3 kHz.
>  
> --
> Tim
> :wq
>  
> On Sep 7, 2013, at 11:19 PM, Harry Romano <hromano at earthlink.net> wrote:
> 
> 
> Is there a way to turn their volume down?

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From kt9ac at ameritech.net  Mon Sep  9 08:53:36 2013
From: kt9ac at ameritech.net (Tony KT9AC)
Date: Mon, 9 Sep 2013 01:53:36 -0700 (PDT)
Subject: [App_rpt-users] announcements
In-Reply-To: <522D14A2.8030508@kuggie.com>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<522CD2C7.8010002@kuggie.com>
	<5C6B91E2-601F-41F2-B15D-6D18BA953FAD@brannan.name>
	<522D14A2.8030508@kuggie.com>
Message-ID: <1378716816.25306.YahooMailNeo@web181101.mail.ne1.yahoo.com>

Kevin,
?Do you mean WAV file recording like ACID does?

? ? Recording and Streaming - sure it does. ?We record everything and it is streamed on Radio Reference.
? ? Contact Brian Burton - KB3ORS for details on this.


Tony


________________________________
 From: Kevin Custer <kuggie at kuggie.com>
To: Buddy Brannan <buddy at brannan.name>; app_rpt-users at ohnosec.org 
Sent: Sunday, September 8, 2013 7:21 PM
Subject: Re: [App_rpt-users] announcements
 


On 9/8/2013 4:38 PM, Buddy Brannan wrote:

I agree that this is a great feature that Xipar has. Also subjective, and I don?t know what they?re doing differently, but I have an easier time with level adjustments with Xipar (the automatic level setting for rxnoise seems to work a bit better), and at least on my radios, audio ?sounds? better, better being kind of relative, yes, I understand that.
I guess it depends on if you are using the audio w-i-d-e-n-i-n-g
    filter rules.? If you are - then they are not relative, but real
    improvements in audio quality.? 
In zipper, the following apply:

rxlpf=x
txlpf=x
txhpf=x
txlpf=x

x can be 0, 1 or 2.
More info here:
http://www.wanrepeater.com/xipar_urd_filters.doc





However, it looks like xipar is missing a couple things that main line acide distro has, I mean besides that acid works with Allstar?s autopatch and dial-in service without additional fiddling. Doesn?t appear Xipar has the provision for recording or streaming from app_rpt.
Recording and Streaming - sure it does.? We record everything and it
    is streamed on Radio Reference.
Contact Brian Burton - KB3ORS for details on this.



Also, it looks like ID pitch and volume are fixed (and a bit lo of my taste),
You have to put in the following in rpt.conf then you can set it
    however you like.
morse = morse
Then, adjust the settings in the [morse] stanza.



and the echo link channel driver didn?t compile last time I tried a beta of it several months ago.
Not sure - I personally don't do Echolink, but it worked the last
    time someone else did a new install and connected to WAN.? 



And i at least don?t really need or want Freepbx. So, until or unless Echolink gets worked out along with streaming/recording, I?m sticking with acid, even in spite of some of xipar?s real advantages. 
That's why there are several distros.

IMHO there are two BIG disadvantages currently with XIPAR - No
    WebTransceiver and no AllMon.?? So, I run two servers with ACID -
    all fixed...

Kevin




_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From harvard5362 at yahoo.com  Mon Sep  9 15:20:21 2013
From: harvard5362 at yahoo.com (C B)
Date: Mon, 9 Sep 2013 08:20:21 -0700 (PDT)
Subject: [App_rpt-users] autopatch when linked
Message-ID: <1378740021.24201.YahooMailNeo@web124503.mail.ne1.yahoo.com>

Hi
?
in trying to figure out a good way to implement auto-patch on a linked system.
?
I keep several repeaters linked full time, some via my existing system with a ALLSTAR node connected to a port on the controller, others via ALLSTAR directly. thus I feel it would be more efficient if 1 patch served all.
?
Can 1 node share its auto-patch with other nodes or do I need to have a separate patch for each node? Also I would like to use a PBX (not sip) extension if possible. 
?
If it is possible to share one auto-patch between nodes could I place a FXO card in the HUB node computer and share that connection?
?
please let me apologize if this has been covered but I could not find it.
?
thank you for your help.
?
Chris
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From cypresstower at yahoo.com  Mon Sep  9 15:26:24 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Mon, 9 Sep 2013 08:26:24 -0700 (PDT)
Subject: [App_rpt-users] announcements
In-Reply-To: <C71C5AA8-5BEC-4D8B-8EC7-81B8DA8BDA74@me.com>
References: <002101ceac5b$5c8710c0$15953240$@net>
	<DC860F48-1D21-4338-B6C1-47BA3A91E48A@me.com>
	<000a01cead12$b36e9b30$1a4bd190$@net>
	<C71C5AA8-5BEC-4D8B-8EC7-81B8DA8BDA74@me.com>
Message-ID: <1378740384.39970.YahooMailNeo@web163602.mail.gq1.yahoo.com>

I find most sysops adjust the audio by ear so that itself adds to the mismatch. ??I like the cop commands, 33-34 to turn on and off the audio.? 
Zipper is good, but as mentioned ACID has its advantages.? One being it still supports ztmonitor.? That's a good tool.? 
Maybe if we all send in a 5 dollar donation once a year, it may in courage the code guys to consider some of the suggestions out there.? 
I'm on my way to kick in right now. 
Johnny
?

________________________________
From: Tim Sawyer <tim.sawyer at me.com>
To: Harry Romano <hromano at earthlink.net> 
Cc: "app_rpt-users at ohnosec.org list" <App_rpt-users at ohnosec.org> 
Sent: Monday, September 9, 2013 1:01 AM
Subject: Re: [App_rpt-users] announcements



Yes it's easy to turn off dis/connect messages. See telemdefault?http://ohnosec.org/drupal/node/103. I like mode 2.?
Echolink levels can be adjusted independently of other audio. In your node stanza you should see the following statements near the top. Uncomment or add them.? 


erxgain=-3 ? ? ? ? ? ? ? ? ? ? ? ? ?; Echolink receive gain adjustment
? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ; Note: Gain is in db-volts (20logVI/VO)
etxgain=3 ? ? ? ? ? ? ? ? ? ? ? ? ? ; Echolink transmit gain adjustment
? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ; Note: Gain is in db-volts (20logVI/VO)

One thing I like to do is add another node to my system just for echolink. That way you can dis/connect it as needed.
--
Tim
:wq

On Sep 8, 2013, at 9:11 PM, Harry Romano <hromano at earthlink.net> wrote:

Hi Tim,
>
>Thanks for the reply . I think I want to shut them off. When I was on the win system the other night there were so many connects and disconnect announcements you could not here the net control. Is It easier to just turn them off instead of ?turning them down?
>
>I have the radios adjusted good for input and output deviation, What I am speaking of is the announcements.? Thanks for the info . Apparently there is a difference when you go to radio tune mode from Cli vs. root.? The cli allows more detailed tuning.
>
>Another issue I have is with Echolink. I personally don?t like Echolink but one of my repeater users does a net with it . The echo link come in at a different volume level than let?s say the web transceiver or another allstar node . both my nodes connect together great but Echolink is defiantly lower . Is there a way to get a volume increase just for Echolink? I was thinking of putting a txmixer = 250 command in the Echolink.conf. But I think that may make everything else get louder too.
>
>
>73's
>
>Harry? KC4RPP
>?
>?
>From:?Tim Sawyer [mailto:tim.sawyer at me.com]?
>Sent:?Sunday, September 08, 2013 1:24 PM
>To:?Harry Romano
>Cc:?app_rpt-users at ohnosec.org
>Subject:?Re: [App_rpt-users] announcements
>
>When you set the TX and RX levels per instructions the announcements will be the correct level.?
>
>
>Please (again) see?http://ohnosec.org/drupal/node/48?for all the gory details of setup with chan_usbradio.?If you use chan_simpleusb the RX procedure is a bit different:?Generate (with a service monitor) a full quieting 1kHz tone at 3kHz dev with no CTCSS. At the OS CLI type radio_tune_menu, adjust the reading for 3kHz.?With a RTCM the procedure is pretty much the same except you have nice indicator LEDs right on the RTCM.
>
>
>Transmit is pretty much the same regardless of the interface. ?Turn of any TX CTCSS. Press *989 to generate the reference test level tone. Then measuring with your service monitor set the dev to 3 kHz.
>
>
>--
>Tim
>:wq
>
>On Sep 7, 2013, at 11:19 PM, Harry Romano <hromano at earthlink.net> wrote:
>
>
>
>Is there a way to turn their volume down?

_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From telesistant at hotmail.com  Mon Sep  9 17:00:08 2013
From: telesistant at hotmail.com (Jim Duuuude)
Date: Mon, 9 Sep 2013 10:00:08 -0700
Subject: [App_rpt-users] autopatch when linked
In-Reply-To: <1378740021.24201.YahooMailNeo@web124503.mail.ne1.yahoo.com>
References: <1378740021.24201.YahooMailNeo@web124503.mail.ne1.yahoo.com>
Message-ID: <BLU180-W17317A7311F6B5121E9B9FB13F0@phx.gbl>

First of all, the Allstar Autopatch  subscription is per SERVER, not NODE. All the nodes
on any server, by inherent design, share autopatch accessibility. 

There is not so much as a SINGLE manufacturer of an FXO card that will provide
sufficient quality, especially in the area of trans-hybrid loss, for radio autopatch
(full duplex) use, without hardware echo cancellation, which is going to be a DARNED
EXPENSIVE CARD!!!! Nor will any ATA!!! Even current FXO cards with H/W echo cancellation
*REALLY* sound CRAPPY when used this way. 

Don't do it. Don't try it. It won't work. That is reality.

Now, for those of you "poised" to press the reply button and share your "wealth" of
advice on how "gee, it works fine for me", I am just going to say, "in this case, just don't".
Trust me. I invented this technology (Zapata Technology/Zaptel/DAHDI). I understand why
it works, and why (in this case) it can't (in great detail).

I certainly acknowledge that there have been a number of really good open S/W echo
cancelers contributed to the Asterisk project, especially the one that David Rowe wrote.
Even it can't "forgive" a poopy hardware FXO implementation (which they ALL are) along
with the SUPER DEMANDING (well in excess of normal telecom) needs of a two-way radio
full duplex autopatch.

I dedicated a vast part of my life, to, amongst other things, making reasonable *FOUR WIRE*
telecom interfaces available to *EVERYONE ON THE PLANET*. Previous to Zapata/Asterisk, we
were all *FORCED* to use 2 wire (standard POTS) for things that *REALLY* required a trunk
(4 wire) for proper quality and control. We can now. And its so incredibly inexpensive. It makes
me smile!! Take advantage of it.

Jim WB6NIL


Date: Mon, 9 Sep 2013 08:20:21 -0700
From: harvard5362 at yahoo.com
To: app_rpt-users at ohnosec.org
Subject: [App_rpt-users] autopatch when linked

Hi in trying to figure out a good way to implement auto-patch on a linked system. I keep several repeaters linked full time, some via my existing system with a ALLSTAR node connected to a port on the controller, others via ALLSTAR directly. thus I feel it would be more efficient if 1 patch served all. Can 1 node share its auto-patch with other nodes or do I need to have a separate patch for each node? Also I would like to use a PBX (not sip) extension if possible.  If it is possible to share one auto-patch between nodes could I place a FXO card in the HUB node computer and share that connection? please let me apologize if this has been covered but I could not find
 it. thank you for your help. Chris
_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 		 	   		  
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From n5uxt at hotmail.com  Mon Sep  9 17:41:28 2013
From: n5uxt at hotmail.com (Angelo Glorioso)
Date: Mon, 9 Sep 2013 12:41:28 -0500
Subject: [App_rpt-users] USB Printer Port
References: <1378740021.24201.YahooMailNeo@web124503.mail.ne1.yahoo.com>
	<BLU180-W17317A7311F6B5121E9B9FB13F0@phx.gbl>
Message-ID: <BLU0-SMTP21327150DAA537B3C3F1D11ED3F0@phx.gbl>

I was wondering if anyone has tried a USB Printer interface with the PRLP interface with Asterisk???

Does ACID address the USB adapter???

7 3 de Angelo

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From n1dot1 at gmail.com  Mon Sep  9 17:46:23 2013
From: n1dot1 at gmail.com (Kenneth Grimard)
Date: Mon, 09 Sep 2013 13:46:23 -0400
Subject: [App_rpt-users] USB Printer Port
In-Reply-To: <BLU0-SMTP21327150DAA537B3C3F1D11ED3F0@phx.gbl>
References: <1378740021.24201.YahooMailNeo@web124503.mail.ne1.yahoo.com>
	<BLU180-W17317A7311F6B5121E9B9FB13F0@phx.gbl>
	<BLU0-SMTP21327150DAA537B3C3F1D11ED3F0@phx.gbl>
Message-ID: <522E096F.8060106@gmail.com>

Hi angelo

I have a radio port running as a prli node. it is using an irlp board 
for logic (ptt and cor) I converted the node to asterisk a while ago.

look here for more info.

http://www.qsl.net/k0kn/

ken n1dot



On 9/9/2013 1:41 PM, Angelo Glorioso wrote:
> I was wondering if anyone has tried a USB Printer interface with the 
> PRLP interface with Asterisk???
> Does ACID address the USB adapter???
> 7 3 de Angelo
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From bbrown at byrg.net  Mon Sep  9 18:16:25 2013
From: bbrown at byrg.net (=?ISO-8859-1?Q?Bob_Brown_=2D_W=D8NQX?=)
Date: Mon, 9 Sep 2013 13:16:25 -0500
Subject: [App_rpt-users] autopatch when linked
In-Reply-To: <BLU180-W17317A7311F6B5121E9B9FB13F0@phx.gbl>
References: <1378740021.24201.YahooMailNeo@web124503.mail.ne1.yahoo.com>
	<BLU180-W17317A7311F6B5121E9B9FB13F0@phx.gbl>
Message-ID: <CA+WM89ZvjPzh8kYN21TwL=SUJqbbW-RFHcqgzhfJXvMivhHw=w@mail.gmail.com>

All,

Listen to Him ( JIM ), He Speaks the Truth.

He has been there, done that and got a few Tattoos and T-shirts, maybe had
the Tats removed, or never had them applied.

But at any rate, he knows best here!



Sm0ke.

--
Thanks in Advance

   Bob Brown, W?NQX

   Kansas City Metro Area

   http://drsm0ke.net

   http://byrg.net

   http://sm0kenet.net

   http://kcdstar.byrg.net

Quis custodiet ipsos custodes?




--


On Mon, Sep 9, 2013 at 12:00 PM, Jim Duuuude <telesistant at hotmail.com>wrote:

> First of all, the Allstar Autopatch  subscription is per SERVER, not NODE.
> All the nodes
> on any server, by inherent design, share autopatch accessibility.
>
> There is not so much as a SINGLE manufacturer of an FXO card that will
> provide
> sufficient quality, especially in the area of trans-hybrid loss, for radio
> autopatch
> (full duplex) use, without hardware echo cancellation, which is going to
> be a DARNED
> EXPENSIVE CARD!!!! Nor will any ATA!!! Even current FXO cards with H/W
> echo cancellation
> *REALLY* sound CRAPPY when used this way.
>
> Don't do it. Don't try it. It won't work. That is reality.
>
> Now, for those of you "poised" to press the reply button and share your
> "wealth" of
> advice on how "gee, it works fine for me", I am just going to say, "in
> this case, just don't".
> Trust me. I invented this technology (Zapata Technology/Zaptel/DAHDI). I
> understand why
> it works, and why (in this case) it can't (in great detail).
>
> I certainly acknowledge that there have been a number of really good open
> S/W echo
> cancelers contributed to the Asterisk project, especially the one that
> David Rowe wrote.
> Even it can't "forgive" a poopy hardware FXO implementation (which they
> ALL are) along
> with the SUPER DEMANDING (well in excess of normal telecom) needs of a
> two-way radio
> full duplex autopatch.
>
> I dedicated a vast part of my life, to, amongst other things, making
> reasonable *FOUR WIRE*
> telecom interfaces available to *EVERYONE ON THE PLANET*. Previous to
> Zapata/Asterisk, we
> were all *FORCED* to use 2 wire (standard POTS) for things that *REALLY*
> required a trunk
> (4 wire) for proper quality and control. We can now. And its so incredibly
> inexpensive. It makes
> me smile!! Take advantage of it.
>
> Jim WB6NIL
>
>
> ------------------------------
> Date: Mon, 9 Sep 2013 08:20:21 -0700
> From: harvard5362 at yahoo.com
> To: app_rpt-users at ohnosec.org
> Subject: [App_rpt-users] autopatch when linked
>
>
> Hi
>
> in trying to figure out a good way to implement auto-patch on a linked
> system.
>
> I keep several repeaters linked full time, some via my existing system
> with a ALLSTAR node connected to a port on the controller, others via
> ALLSTAR directly. thus I feel it would be more efficient if 1 patch served
> all.
>
> Can 1 node share its auto-patch with other nodes or do I need to have a
> separate patch for each node? Also I would like to use a PBX (not sip)
> extension if possible.
>
> If it is possible to share one auto-patch between nodes could I place a
> FXO card in the HUB node computer and share that connection?
>
> please let me apologize if this has been covered but I could not find it.
>
> thank you for your help.
>
> Chris
>
> _______________________________________________ App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
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From yokshs at gmail.com  Mon Sep  9 22:45:22 2013
From: yokshs at gmail.com (K&R Yoksh)
Date: Mon, 9 Sep 2013 17:45:22 -0500
Subject: [App_rpt-users] USB Printer Port
Message-ID: <CAJwRzNN9526QWxbhUa36F_G=Bt=qwrxoAt8mbLD21PJQtG5kww@mail.gmail.com>

I think that Angelo is speaking about USB->Parallel adapters being used on
computers that lack a physical parallel port.

I believe that this question has come up before, and the answer was NO, it
doesn't work.

I recommended using a PCI parallel port card, but Angelo is working with a
laptop.

73,

Kyle
K0KN
Olathe, KS
Allstar 2210-2219

--- Original Message ---

I was wondering if anyone has tried a USB Printer interface with the PRLP
interface with Asterisk???

Does ACID address the USB adapter???

7 3 de Angelo
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From mike at midnighteng.com  Mon Sep  9 23:03:33 2013
From: mike at midnighteng.com (mike at midnighteng.com)
Date: Mon, 09 Sep 2013 16:03:33 -0700
Subject: [App_rpt-users] =?utf-8?q?USB_Printer_Port?=
Message-ID: <20130909160333.71befee5dbd13c5325dd1a521b4e73ee.fdd7289895.wbe@email06.secureserver.net>

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From vk3jed at vkradio.com  Tue Sep 10 09:29:23 2013
From: vk3jed at vkradio.com (Tony Langdon)
Date: Tue, 10 Sep 2013 19:29:23 +1000
Subject: [App_rpt-users] USB Printer Port
In-Reply-To: <BLU0-SMTP21327150DAA537B3C3F1D11ED3F0@phx.gbl>
References: <1378740021.24201.YahooMailNeo@web124503.mail.ne1.yahoo.com>
	<BLU180-W17317A7311F6B5121E9B9FB13F0@phx.gbl>
	<BLU0-SMTP21327150DAA537B3C3F1D11ED3F0@phx.gbl>
Message-ID: <522EE673.3060704@vkradio.com>

On 10/09/13 3:41 AM, Angelo Glorioso wrote:
> I was wondering if anyone has tried a USB Printer interface with the 
> PRLP interface with Asterisk???
> Does ACID address the USB adapter???
USB parallel ports do not support the IRLP board.  This is because the 
IRLP board uses direct manipulation of the I/O lines, rather than the 
packetised data format the USB adapters expect.

-- 
73 de Tony VK3JED/VK3IRL
http://vkradio.com

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From n5uxt at hotmail.com  Tue Sep 10 14:35:31 2013
From: n5uxt at hotmail.com (Angelo Glorioso)
Date: Tue, 10 Sep 2013 09:35:31 -0500
Subject: [App_rpt-users] USB Printer Port
References: <20130909160333.71befee5dbd13c5325dd1a521b4e73ee.fdd7289895.wbe@email06.secureserver.net>
Message-ID: <BLU0-SMTP20900175D8A203CA5D25BA0ED380@phx.gbl>

Mike,

 I would think so. I am not a Linux person myself, but I guess it is how the Centos addresses I/O request.

7 3 de Angelo

 
  ----- Original Message ----- 
  From: mike at midnighteng.com 
  To: app_rpt-users at ohnosec.org 
  Sent: Monday, September 09, 2013 6:03 PM
  Subject: Re: [App_rpt-users] USB Printer Port



  Should be no reason that a USB/PP would not work with this if it is properly addressed (i.e 378,278,3bc) in your configuration.



  While I have not used them with apt-rpt specifically, I have used them with controller software that I wrote years ago.


  ...mike/kb8jnm





    -------- Original Message --------
    Subject: [App_rpt-users] USB Printer Port
    From: "K&R Yoksh" <yokshs at gmail.com>
    Date: Mon, September 09, 2013 6:45 pm
    To: app_rpt-users at ohnosec.org



    I think that Angelo is speaking about USB->Parallel adapters being used on computers that lack a physical parallel port.

    I believe that this question has come up before, and the answer was NO, it doesn't work.

    I recommended using a PCI parallel port card, but Angelo is working with a laptop.

    73,

    Kyle
    K0KN
    Olathe, KS
    Allstar 2210-2219

    --- Original Message ---

    I was wondering if anyone has tried a USB Printer interface with the PRLP interface with Asterisk???

    Does ACID address the USB adapter???

    7 3 de Angelo


----------------------------------------------------------------------------
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    App_rpt-users mailing list
    App_rpt-users at ohnosec.org
    http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



------------------------------------------------------------------------------


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From andy at ple.org  Tue Sep 10 23:38:47 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Tue, 10 Sep 2013 18:38:47 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
Message-ID: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>

Good

sometimes, i get distorted audio locally on the repeat and also on simple announcements like the time announcement. If and when I get that distorted audio it kind of sounds as if the audio stream is sampled i 20ms increments and every other piece is left out. This happens only every now and then, while most of the time the audio is absolutely perfect. 

As hardware is a nondescript 2GHz dual core, (all-star uses about 15-25% CPU), so it should not be a speed problem. Two URI?s are connected to this PC, one for the public all-star link, the other to a private GMRS node, 

Tried two different radios, so it should not be the radio either, unless two unrelated GM300s have exactly the same problem (For the time being I?ll go under the assumption its not the radio) I don?t know yet if that ever happens on the TCP end, but I doubt it. 

Does anyone have *ANY* idea what that could be?
--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

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From jrorke at cogeco.ca  Wed Sep 11 11:51:15 2013
From: jrorke at cogeco.ca (Jon Rorke)
Date: Wed, 11 Sep 2013 07:51:15 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
Message-ID: <52305933.2040001@cogeco.ca>

I have seen this before.
What is the TX voice setting on y our node? If its any where near 900 
then it may be trying to drive your TX but is running out of room.
Do you have TX boost turned on in USBradio.conf?

> hdwtype=0
> rxboost=0
> txboost=1

If not turn it on then retune the TX voice. If you can TX voice near 500 
(Mid scale) then you have a better chance of not having the audio 
distort on some peaks.
The same would be true if your TX voice setting is near 100 to 200 and 
you are over driving the TX a little. In this case then turn off TX 
boost in USBradio and retune.

Give this a try ans see if this helps.

Jon VA3RQ

On 9/10/2013 7:38 PM, Andreas Pleschutznig wrote:
> Good
>
> sometimes, i get distorted audio locally on the repeat and also on 
> simple announcements like the time announcement. If and when I get 
> that distorted audio it kind of sounds as if the audio stream is 
> sampled i 20ms increments and every other piece is left out. This 
> happens only every now and then, while most of the time the audio is 
> absolutely perfect.
>
> As hardware is a nondescript 2GHz dual core, (all-star uses about 
> 15-25% CPU), so it should not be a speed problem. Two URI's are 
> connected to this PC, one for the public all-star link, the other to a 
> private GMRS node,
>
> Tried two different radios, so it should not be the radio either, 
> unless two unrelated GM300s have exactly the same problem (For the 
> time being I'll go under the assumption its not the radio) I don't 
> know yet if that ever happens on the TCP end, but I doubt it.
>
> Does anyone have *ANY* idea what that could be?
> *--*
> *Andreas Pleschutznig, *Cell: (832) 633-7817, Sent: MacBook
>  KA5PLE, Allstar: 29841, Echolonk: 884823
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From cypresstower at yahoo.com  Wed Sep 11 16:07:12 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Wed, 11 Sep 2013 09:07:12 -0700 (PDT)
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <52305933.2040001@cogeco.ca>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
Message-ID: <1378915632.82056.YahooMailNeo@web163601.mail.gq1.yahoo.com>

Just another perspective on audio.
I too had a similar experience with occasional?distortion.? So I got test gear and checked the audio.? 
It was right on, but found the receiver was drifting off frequency?from temp changes.? It's been pointed 
out to me, a very large % of?times it's a hardware failure, as in my case.? Test gear makes me happy J
Please; Let us know what you find out.?
JK?



From: Jon Rorke <jrorke at cogeco.ca>
To: app_rpt-users at ohnosec.org 
Sent: Wednesday, September 11, 2013 7:51 AM
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar



I have seen this before. 
What is the TX voice setting on y our node? If its any where near 900 then it may be trying to drive your TX but is running out of room.
Do you have TX boost turned on in USBradio.conf? 


hdwtype=0?? ?
>rxboost=0
>txboost=1
If not turn it on then retune the TX voice. If you can TX voice near 500 (Mid scale) then you have a better chance of not having the audio distort on some peaks.
The same would be true if your TX voice setting is near 100 to 200 and you are over driving the TX a little. In this case then turn off TX boost in USBradio and retune. 

Give this a try ans see if this helps.

Jon VA3RQ


On 9/10/2013 7:38 PM, Andreas Pleschutznig wrote:

Good 
>
>
>sometimes, i get distorted audio locally on the repeat and also on simple announcements like the time announcement. If and when I get that distorted audio it kind of sounds as if the audio stream is sampled i 20ms increments and every other piece is left out. This happens only every now and then, while most of the time the audio is absolutely perfect.?
>
>
>As hardware is a nondescript 2GHz dual core, (all-star uses about 15-25% CPU), so it should not be a speed problem. Two URI?s are connected to this PC, one for the public all-star link, the other to a private GMRS node,?
>
>
>Tried two different radios, so it should not be the radio either, unless two unrelated GM300s have exactly the same problem (For the time being I?ll go under the assumption its not the radio) I don?t know yet if that ever happens on the TCP end, but I doubt it.?
>
>
>Does anyone have *ANY* idea what that could be?
>
>--
>Andreas Pleschutznig,?Cell: (832) 633-7817, Sent: MacBook
>?KA5PLE, Allstar: 29841, Echolonk: 884823
>
> 
>
>_______________________________________________
App_rpt-users mailing list App_rpt-users at ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 

_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From andy at ple.org  Wed Sep 11 16:20:32 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Wed, 11 Sep 2013 11:20:32 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <52305933.2040001@cogeco.ca>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
Message-ID: <80B34670-B925-4472-9692-60B46A306820@ple.org>

Hi Jon

I do not believe this is distorted audio because it is too low or too high. The audio has been measured in with the help of a service monitor and I just checked it. Both channels are set to values around 100. Also distorted because it is set too high or too low sounds different than what I am seeing here. 

Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time. 

 
--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 6:51 AM, Jon Rorke <jrorke at cogeco.ca> wrote:

> I have seen this before. 
> What is the TX voice setting on y our node? If its any where near 900 then it may be trying to drive your TX but is running out of room.
> Do you have TX boost turned on in USBradio.conf? 
> 
>> hdwtype=0    
>> rxboost=0
>> txboost=1
> 
> If not turn it on then retune the TX voice. If you can TX voice near 500 (Mid scale) then you have a better chance of not having the audio distort on some peaks.
> The same would be true if your TX voice setting is near 100 to 200 and you are over driving the TX a little. In this case then turn off TX boost in USBradio and retune. 
> 
> Give this a try ans see if this helps.
> 
> Jon VA3RQ
> 
> On 9/10/2013 7:38 PM, Andreas Pleschutznig wrote:
>> Good
>> 
>> sometimes, i get distorted audio locally on the repeat and also on simple announcements like the time announcement. If and when I get that distorted audio it kind of sounds as if the audio stream is sampled i 20ms increments and every other piece is left out. This happens only every now and then, while most of the time the audio is absolutely perfect. 
>> 
>> As hardware is a nondescript 2GHz dual core, (all-star uses about 15-25% CPU), so it should not be a speed problem. Two URI?s are connected to this PC, one for the public all-star link, the other to a private GMRS node, 
>> 
>> Tried two different radios, so it should not be the radio either, unless two unrelated GM300s have exactly the same problem (For the time being I?ll go under the assumption its not the radio) I don?t know yet if that ever happens on the TCP end, but I doubt it. 
>> 
>> Does anyone have *ANY* idea what that could be?
>> --
>> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>>  KA5PLE, Allstar: 29841, Echolonk: 884823
>> 
>> 
>> 
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From andy at ple.org  Wed Sep 11 16:23:17 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Wed, 11 Sep 2013 11:23:17 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <1378915632.82056.YahooMailNeo@web163601.mail.gq1.yahoo.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<1378915632.82056.YahooMailNeo@web163601.mail.gq1.yahoo.com>
Message-ID: <2F05909A-7FCE-4B93-B20E-6808D75FFB19@ple.org>

Hi Johnny,

I have tried that one already. I have exactly the same effect on two different radios, in two different bands. As much as I have to replace one of the radios because it is on its way out (slightly of center) in this case it does not seem to be a specific radio?s problem. 

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 11:07 AM, Johnny Keeker <cypresstower at yahoo.com> wrote:

> Just another perspective on audio.
> I too had a similar experience with occasional distortion.  So I got test gear and checked the audio. 
> It was right on, but found the receiver was drifting off frequency from temp changes.  It's been pointed
> out to me, a very large % of times it's a hardware failure, as in my case.  Test gear makes me happy J
> Please; Let us know what you find out. 
> JK 
>  
> From: Jon Rorke <jrorke at cogeco.ca>
> To: app_rpt-users at ohnosec.org 
> Sent: Wednesday, September 11, 2013 7:51 AM
> Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar
> 
> I have seen this before. 
> What is the TX voice setting on y our node? If its any where near 900 then it may be trying to drive your TX but is running out of room.
> Do you have TX boost turned on in USBradio.conf? 
> 
>> hdwtype=0    
>> rxboost=0
>> txboost=1
> 
> If not turn it on then retune the TX voice. If you can TX voice near 500 (Mid scale) then you have a better chance of not having the audio distort on some peaks.
> The same would be true if your TX voice setting is near 100 to 200 and you are over driving the TX a little. In this case then turn off TX boost in USBradio and retune. 
> 
> Give this a try ans see if this helps.
> 
> Jon VA3RQ
> 
> On 9/10/2013 7:38 PM, Andreas Pleschutznig wrote:
>> Good
>> 
>> sometimes, i get distorted audio locally on the repeat and also on simple announcements like the time announcement. If and when I get that distorted audio it kind of sounds as if the audio stream is sampled i 20ms increments and every other piece is left out. This happens only every now and then, while most of the time the audio is absolutely perfect. 
>> 
>> As hardware is a nondescript 2GHz dual core, (all-star uses about 15-25% CPU), so it should not be a speed problem. Two URI?s are connected to this PC, one for the public all-star link, the other to a private GMRS node, 
>> 
>> Tried two different radios, so it should not be the radio either, unless two unrelated GM300s have exactly the same problem (For the time being I?ll go under the assumption its not the radio) I don?t know yet if that ever happens on the TCP end, but I doubt it. 
>> 
>> Does anyone have *ANY* idea what that could be?
>> --
>> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>>  KA5PLE, Allstar: 29841, Echolonk: 884823
>> 
>> 
>> 
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From kuggie at kuggie.com  Wed Sep 11 16:45:27 2013
From: kuggie at kuggie.com (Kevin Custer)
Date: Wed, 11 Sep 2013 12:45:27 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <80B34670-B925-4472-9692-60B46A306820@ple.org>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
Message-ID: <52309E27.5080209@kuggie.com>

On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:
>
>
> Best I can describe the effect is that the audio is time sliced in 
> 20ms packets and every other packet is missing, left out. and it does 
> not happen when audio is especially loud or quiet. Sometimes it 
> happens on the time announcements, which are absolutely perfect 99% of 
> the time.

Sounds to me like a computer with too little RAM or too little CPU 
horsepower.

When you run the top command, is it using swap memory?  If so, that's 
bad...and you need more RAM.

Kevin

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From keith at handscombe.co.uk  Wed Sep 11 17:34:47 2013
From: keith at handscombe.co.uk (Keith@handscombe.co.uk)
Date: Wed, 11 Sep 2013 18:34:47 +0100
Subject: [App_rpt-users] Not on status screen status.allstar.org
Message-ID: <7EB870A1-4F56-4B43-B3E8-6B2E1AD3B16A@handscombe.co.uk>

Evening all from the UK. I run node 2498 and all is running fine. I check the status screen on Allstars site and my node number is not listed.
When I try and login to the portal the syntax I get back is password is still outstanding so I am unable to login to portal or try reset password as its refreshes to the same screen in a loop.
I have tried IE9 and Firefox browser on different pc's both report the same fault.

On my node I can still ping stats.allstar.org and my node gets a reply

Any help would be great

Kind Regards

Keith Handscombe 
Tel:  07940 849210

From george at dyb.com  Wed Sep 11 17:44:14 2013
From: george at dyb.com (George Csahanin)
Date: Wed, 11 Sep 2013 12:44:14 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <52309E27.5080209@kuggie.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org><52305933.2040001@cogeco.ca><80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
Message-ID: <657AC31609934FC08D547E3EF8FFBEFA@georgecPCOMVC>

I have that exact problem if I use DSP, moved to COS and PTT on DB25 and it stops. Did this in every case I tried, though the only platform I used was a D945 or D201 Intel board.

Ran 1GB of RAM and CPU is whatever a D945 is. 

GeorgeC
W2DB
2360


From: Kevin Custer 
Sent: Wednesday, September 11, 2013 11:45 AM
To: Andreas Pleschutznig 
Cc: app_rpt-users at ohnosec.org 
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar

On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:



  Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time. 


Sounds to me like a computer with too little RAM or too little CPU horsepower.

When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.

Kevin

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From ars.w5omr at gmail.com  Wed Sep 11 17:47:56 2013
From: ars.w5omr at gmail.com (Geoff Edmonson)
Date: Wed, 11 Sep 2013 12:47:56 -0500
Subject: [App_rpt-users] Not on status screen status.allstar.org
Message-ID: <uqgi8y9u8uxpiswo4dxtwt0r.1378921676936@email.android.com>

If you're having password issues, change to something with only letters and numbers. @#$%&*-+()!, etc are considered "special characters".

I had the same issue.

Geoff/W5OMR

From bdboyle at bdboyle.com  Wed Sep 11 17:56:24 2013
From: bdboyle at bdboyle.com (Bryan D. Boyle)
Date: Wed, 11 Sep 2013 13:56:24 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <657AC31609934FC08D547E3EF8FFBEFA@georgecPCOMVC>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<657AC31609934FC08D547E3EF8FFBEFA@georgecPCOMVC>
Message-ID: <605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>

DSP takes a significant amount of horsepower.  Ya don't get something for nothing; reading a hardware status bit or two is a lot less intensive at the expense of additional wiring and configuration statements in the config files than decoding the stream, deriving the signals, etc.  any overhead in doing so is bound to have an effect on other streams going on at that time.

and we humans don't process breaks in audio as we do with visual perceptions.

--
Bryan
Sent from my iPhone 5...small
keyboard, big fingers...please
forgive misspellings...



On Sep 11, 2013, at 13:44, "George Csahanin" <george at dyb.com> wrote:

> I have that exact problem if I use DSP, moved to COS and PTT on DB25 and it stops. Did this in every case I tried, though the only platform I used was a D945 or D201 Intel board.
>  
> Ran 1GB of RAM and CPU is whatever a D945 is.
>  
> GeorgeC
> W2DB
> 2360
>  
>  
> From: Kevin Custer
> Sent: Wednesday, September 11, 2013 11:45 AM
> To: Andreas Pleschutznig
> Cc: app_rpt-users at ohnosec.org
> Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar
>  
> On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:
>>  
>>  
>> Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time. 
> 
> Sounds to me like a computer with too little RAM or too little CPU horsepower.
> 
> When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.
> 
> Kevin
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From andy at ple.org  Wed Sep 11 18:00:53 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Wed, 11 Sep 2013 13:00:53 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <52309E27.5080209@kuggie.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
Message-ID: <A94F9FC9-3489-4935-8458-732201FF603E@ple.org>

4GB should be enough me more and 2x2GHz CPUs should be more than enough for asterisk. AT least it was in the past.

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 11:45 AM, Kevin Custer <kuggie at kuggie.com> wrote:

> On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:
>> 
>> 
>> Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time. 
> 
> Sounds to me like a computer with too little RAM or too little CPU horsepower.
> 
> When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.
> 
> Kevin
> 

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From andy at ple.org  Wed Sep 11 18:06:20 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Wed, 11 Sep 2013 13:06:20 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<657AC31609934FC08D547E3EF8FFBEFA@georgecPCOMVC>
	<605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>
Message-ID: <4FB428D6-B4D1-4CA6-8528-8C9B8CF01844@ple.org>

George, Bryan,

the problem *I* have is not limited to a repeat. it does happen when on simple time announcements. At the time when it happens the system is at less than 35% CPU, so I kind of doubt this is a problem in the CPU being too slow. 

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 12:56 PM, Bryan D. Boyle <bdboyle at bdboyle.com> wrote:

> DSP takes a significant amount of horsepower.  Ya don't get something for nothing; reading a hardware status bit or two is a lot less intensive at the expense of additional wiring and configuration statements in the config files than decoding the stream, deriving the signals, etc.  any overhead in doing so is bound to have an effect on other streams going on at that time.
> 
> and we humans don't process breaks in audio as we do with visual perceptions.
> 
> --
> Bryan
> Sent from my iPhone 5...small
> keyboard, big fingers...please
> forgive misspellings...
> 
> 
> 
> On Sep 11, 2013, at 13:44, "George Csahanin" <george at dyb.com> wrote:
> 
>> I have that exact problem if I use DSP, moved to COS and PTT on DB25 and it stops. Did this in every case I tried, though the only platform I used was a D945 or D201 Intel board.
>>  
>> Ran 1GB of RAM and CPU is whatever a D945 is.
>>  
>> GeorgeC
>> W2DB
>> 2360
>>  
>>  
>> From: Kevin Custer
>> Sent: Wednesday, September 11, 2013 11:45 AM
>> To: Andreas Pleschutznig
>> Cc: app_rpt-users at ohnosec.org
>> Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar
>>  
>> On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:
>>>  
>>>  
>>> Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time. 
>> 
>> Sounds to me like a computer with too little RAM or too little CPU horsepower.
>> 
>> When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.
>> 
>> Kevin
>> 
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From andy at ple.org  Wed Sep 11 18:07:43 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Wed, 11 Sep 2013 13:07:43 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <52309E27.5080209@kuggie.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
Message-ID: <B1CFF4C2-2190-433F-A0A1-87334240A0BB@ple.org>

Hi Kevin

when it does that the system is at less than 35% CPU and still has about 3GB of RAM open (this is on a 4GB system) 

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 11:45 AM, Kevin Custer <kuggie at kuggie.com> wrote:

> On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:
>> 
>> 
>> Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time. 
> 
> Sounds to me like a computer with too little RAM or too little CPU horsepower.
> 
> When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.
> 
> Kevin
> 

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From N1XBM at amsat.org  Wed Sep 11 19:14:50 2013
From: N1XBM at amsat.org (Robert Newberry)
Date: Wed, 11 Sep 2013 15:14:50 -0400
Subject: [App_rpt-users] Lots of questions
In-Reply-To: <CAM09xEbTCCp9LfwqiFz2xU=Y=Sg+iZFAhukfMaLyVpz5AuKhHQ@mail.gmail.com>
References: <CAM09xEbTCCp9LfwqiFz2xU=Y=Sg+iZFAhukfMaLyVpz5AuKhHQ@mail.gmail.com>
Message-ID: <CAM09xEaBYxEgUR96N6JMN8-MGs-zw-vcmimdP-ki=xgdRxw=XA@mail.gmail.com>

I have a mastr ii repeater. I'd like to use app-rpt as the controller for
the repeater. I'd like to be able to run echolink off the repeater. I also
have a URI interface. I'd also like to after one repeater is on the air and
available on echolink be able to do the same thing to another repeater to
make a linked network via IP

My questions are can echolink and app_rpt be run at the same time? At the
same time providing repeater control and echolink service.

Has anyone had any success using a raspberry pi with app_rpt and a URI? I
have a laptop I could use, but the pi is an attractive solution.

I'd also like to add I'm new to linux. I'm OK on computers and networking I
also work in land mobile radio so the RF end of it I'm good.

Apparare Scientior
Paratus Communicare
-N1XBM
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From kuggie at kuggie.com  Wed Sep 11 20:48:32 2013
From: kuggie at kuggie.com (Kevin Custer)
Date: Wed, 11 Sep 2013 16:48:32 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <A94F9FC9-3489-4935-8458-732201FF603E@ple.org>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<A94F9FC9-3489-4935-8458-732201FF603E@ple.org>
Message-ID: <5230D720.1090802@kuggie.com>

On 9/11/2013 2:00 PM, Andreas Pleschutznig wrote:
> 4GB should be enough me more and 2x2GHz CPUs should be more than 
> enough for asterisk. AT least it was in the past.

Yes - it certainly should.

How about your Internet connection?

Kevin

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From kuggie at kuggie.com  Wed Sep 11 20:46:37 2013
From: kuggie at kuggie.com (Kevin Custer)
Date: Wed, 11 Sep 2013 16:46:37 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<657AC31609934FC08D547E3EF8FFBEFA@georgecPCOMVC>
	<605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>
Message-ID: <5230D6AD.5070900@kuggie.com>

On 9/11/2013 1:56 PM, Bryan D. Boyle wrote:
>
>
> and we humans don't process breaks in audio as we do with visual 
> perceptions.


There is GREAT truth to this statement.

Motorola, back before the era the MICOR radio set was being developed, 
invested great time and money in this very issue.

It is my understanding, from talking to the inventor and several 
engineers that worked at Motorola during the MICOR era that the 
circuitry, which ultimately resulted in the famous M6709/M7716 squelch 
chip, was originally designed for receivers used in NASA space missions 
(moon).  Obviously, it is crucial the circuit opened squelch under the 
slightest amount of quieting.   In addition, it was discovered that the 
human brain had a difficult time understanding words that are broken by 
silence; it is much easier if the unintelligent space is filled with 
noise.  For this reason, what became known as the MICOR squelch was a 
circuit that accomplished this task, as well as having the ability to 
close quickly if the recovered audio was considerably noiseless.

Seasoned operators can put together enough recovered audio to make sense 
out of a broken transmission, and people didn't have to listen to a 
squelch burst if the recovered transmission was fairly quiet.

This circuitry and theory can be reviewed in detail by going to the 
USPTO and do a patent number search;
3628058


Kevin - WJ8G


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From n3fe at repeater.net  Wed Sep 11 21:01:48 2013
From: n3fe at repeater.net (Corey Dean)
Date: Wed, 11 Sep 2013 17:01:48 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <52309E27.5080209@kuggie.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
Message-ID: <4BCC91CBCFD66C4489B4BD3233140C3E048596B6D2C2@exchange.mail.repeater.net>

Any chance you are using an AMD processor?  If you tx tone and your voice are more than 999 combined, that could also be the problem.

Corey N3FE

From: app_rpt-users-bounces at ohnosec.org [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Kevin Custer
Sent: Wednesday, September 11, 2013 12:45 PM
To: Andreas Pleschutznig
Cc: app_rpt-users at ohnosec.org
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar

On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:


Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time.

Sounds to me like a computer with too little RAM or too little CPU horsepower.

When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.

Kevin


--
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From andy at ple.org  Wed Sep 11 22:26:26 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Wed, 11 Sep 2013 17:26:26 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <5230D720.1090802@kuggie.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<A94F9FC9-3489-4935-8458-732201FF603E@ple.org>
	<5230D720.1090802@kuggie.com>
Message-ID: <26C787A0-3D15-4D71-9AD1-0C125EFB52B3@ple.org>

Hi Kevin, the internet connection if anything is too fast as well. Besides, even if I had no internet connection that should noyt have any impact on time announcements, right?

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 3:48 PM, Kevin Custer <kuggie at kuggie.com> wrote:

> On 9/11/2013 2:00 PM, Andreas Pleschutznig wrote:
>> 4GB should be enough me more and 2x2GHz CPUs should be more than enough for asterisk. AT least it was in the past.
> 
> Yes - it certainly should.
> 
> How about your Internet connection?
> 
> Kevin
> 

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From andy at ple.org  Wed Sep 11 22:31:38 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Wed, 11 Sep 2013 17:31:38 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <4BCC91CBCFD66C4489B4BD3233140C3E048596B6D2C2@exchange.mail.repeater.net>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<4BCC91CBCFD66C4489B4BD3233140C3E048596B6D2C2@exchange.mail.repeater.net>
Message-ID: <B6385102-C597-4035-921D-DD29339DF61E@ple.org>

Huh? Now you see me stumped. I can see some badly designed math library having a problem with that, but what does that have to do with the processor. I am not discounting the statement, just trying to gather information, and the statement does not just come through the filter I have in my brain as to makes sense.

As far as the CPU goes, yes I do have an AMD chip:

[root at s02 proc]# cat /proc/cpuinfo
processor	: 0
vendor_id	: AuthenticAMD
cpu family	: 20
model		: 1
model name	: AMD E-350 Processor
stepping	: 0
cpu MHz		: 1597.148
cache size	: 512 KB
physical id	: 0
siblings	: 2
core id		: 0
cpu cores	: 2
apicid		: 0
fdiv_bug	: no
hlt_bug		: no
f00f_bug	: no
coma_bug	: no
fpu		: yes
fpu_exception	: yes
cpuid level	: 6
wp		: yes
flags		: fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
bogomips	: 3194.29

processor	: 1
vendor_id	: AuthenticAMD
cpu family	: 20
model		: 1
model name	: AMD E-350 Processor
stepping	: 0
cpu MHz		: 1597.148
cache size	: 512 KB
physical id	: 0
siblings	: 2
core id		: 1
cpu cores	: 2
apicid		: 1
fdiv_bug	: no
hlt_bug		: no
f00f_bug	: no
coma_bug	: no
fpu		: yes
fpu_exception	: yes
cpuid level	: 6
wp		: yes
flags		: fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
bogomips	: 3194.88

[root at s02 proc]# 

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 4:01 PM, Corey Dean <n3fe at repeater.net> wrote:

> Any chance you are using an AMD processor?  If you tx tone and your voice are more than 999 combined, that could also be the problem.
>  
> Corey N3FE
>  
> From: app_rpt-users-bounces at ohnosec.org [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Kevin Custer
> Sent: Wednesday, September 11, 2013 12:45 PM
> To: Andreas Pleschutznig
> Cc: app_rpt-users at ohnosec.org
> Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar
>  
> On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:
>  
>  
> Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time.
> 
> Sounds to me like a computer with too little RAM or too little CPU horsepower.
> 
> When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.
> 
> Kevin
> 
> 
> -- 
> This message was scanned and is believed to be clean. 
> Click here to report this message as spam.

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From n3fe at repeater.net  Wed Sep 11 23:02:41 2013
From: n3fe at repeater.net (Corey Dean)
Date: Wed, 11 Sep 2013 19:02:41 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <B6385102-C597-4035-921D-DD29339DF61E@ple.org>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<4BCC91CBCFD66C4489B4BD3233140C3E048596B6D2C2@exchange.mail.repeater.net>
	<B6385102-C597-4035-921D-DD29339DF61E@ple.org>
Message-ID: <F7CA14B6-B9C0-44C9-AF53-48FA72A9CA95@repeater.net>

I have nothing but problems with an amd processor that I was using as my first node.  A lot of it was audio related as well.  I can't remember if it was Steve or Jim that ask me the very same question.  I ended up trying an intel CPU on a pretty Low end machine and all my problems went away.  I have built many nodes since for both myself and others and have never had that problem show up again.

Corey n3fe
Sent from my iPhone

On Sep 11, 2013, at 6:39 PM, "Andreas Pleschutznig" <andy at ple.org<mailto:andy at ple.org>> wrote:

Huh? Now you see me stumped. I can see some badly designed math library having a problem with that, but what does that have to do with the processor. I am not discounting the statement, just trying to gather information, and the statement does not just come through the filter I have in my brain as to makes sense.

As far as the CPU goes, yes I do have an AMD chip:

[root at s02 proc]# cat /proc/cpuinfo
processor : 0
vendor_id : AuthenticAMD
cpu family : 20
model : 1
model name : AMD E-350 Processor
stepping : 0
cpu MHz : 1597.148
cache size : 512 KB
physical id : 0
siblings : 2
core id : 0
cpu cores : 2
apicid : 0
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 6
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
bogomips : 3194.29

processor : 1
vendor_id : AuthenticAMD
cpu family : 20
model : 1
model name : AMD E-350 Processor
stepping : 0
cpu MHz : 1597.148
cache size : 512 KB
physical id : 0
siblings : 2
core id : 1
cpu cores : 2
apicid : 1
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 6
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
bogomips : 3194.88

[root at s02 proc]#

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 4:01 PM, Corey Dean <n3fe at repeater.net<mailto:n3fe at repeater.net>> wrote:

Any chance you are using an AMD processor?  If you tx tone and your voice are more than 999 combined, that could also be the problem.

Corey N3FE

From: app_rpt-users-bounces at ohnosec.org<mailto:app_rpt-users-bounces at ohnosec.org> [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Kevin Custer
Sent: Wednesday, September 11, 2013 12:45 PM
To: Andreas Pleschutznig
Cc: app_rpt-users at ohnosec.org<mailto:app_rpt-users at ohnosec.org>
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar

On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:


Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time.

Sounds to me like a computer with too little RAM or too little CPU horsepower.

When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.

Kevin


--
This message was scanned and is believed to be clean.
Click here to report this message as spam.<http://simba.repeater.net:8080/cgi-bin/learn-msg.cgi?id=A3CC12223.A96FE>

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From kb4fxc at inttek.net  Wed Sep 11 23:06:48 2013
From: kb4fxc at inttek.net (David McGough)
Date: Wed, 11 Sep 2013 19:06:48 -0400 (EDT)
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <B6385102-C597-4035-921D-DD29339DF61E@ple.org>
Message-ID: <Pine.LNX.4.44.1309111838300.15898-100000@goliath.inttek.net>


It never fails to amaze me how AMD CPU's get blamed for various
issues--and the actual CPU's are almost never at fault! And, yes, AMD and
Intel have both had an occasional "oops" impacting a specific CPU series.

In my experience, most of these "blame" issues arise from really 
poorly designed/buggy motherboards. Unfortunately, in practice, more of 
these bad boards seem to be found in low-end AMD boxes, where system 
builders are cutting every cost corner.

With a good quality MB, AMD and Intel CPU's both work great and exhibit 
similar reliability/stability....And, in case you're wondering, I run a 
Co-Lo center and have a few racks full of AMD boards! I measure up-times 
in years.

73, David KB4FXC


On Wed, 11 Sep 2013, Andreas Pleschutznig wrote:

> Huh? Now you see me stumped. I can see some badly designed math library 
having a problem with that, but what does that have to do with the 
processor. I am not discounting the statement, just trying to gather 
information, and the statement does not just come through the filter I 
have in my brain as to makes sense.
> 
> As far as the CPU goes, yes I do have an AMD chip:
> 
> [root at s02 proc]# cat /proc/cpuinfo
> processor	: 0
> vendor_id	: AuthenticAMD
> cpu family	: 20
> model		: 1
> model name	: AMD E-350 Processor
> stepping	: 0
> cpu MHz		: 1597.148
> cache size	: 512 KB
> physical id	: 0
> siblings	: 2
> core id		: 0
> cpu cores	: 2
> apicid		: 0
> fdiv_bug	: no
> hlt_bug		: no
> f00f_bug	: no
> coma_bug	: no
> fpu		: yes
> fpu_exception	: yes
> cpuid level	: 6
> wp		: yes
> flags		: fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
> bogomips	: 3194.29
> 
> processor	: 1
> vendor_id	: AuthenticAMD
> cpu family	: 20
> model		: 1
> model name	: AMD E-350 Processor
> stepping	: 0
> cpu MHz		: 1597.148
> cache size	: 512 KB
> physical id	: 0
> siblings	: 2
> core id		: 1
> cpu cores	: 2
> apicid		: 1
> fdiv_bug	: no
> hlt_bug		: no
> f00f_bug	: no
> coma_bug	: no
> fpu		: yes
> fpu_exception	: yes
> cpuid level	: 6
> wp		: yes
> flags		: fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
> bogomips	: 3194.88
> 
> [root at s02 proc]# 
> 
> --
> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>  KA5PLE, Allstar: 29841, Echolonk: 884823
> 
> On Sep 11, 2013, at 4:01 PM, Corey Dean <n3fe at repeater.net> wrote:
> 
> > Any chance you are using an AMD processor?  If you tx tone and your voice are more than 999 combined, that could also be the problem.
> >  
> > Corey N3FE
> >  
> > From: app_rpt-users-bounces at ohnosec.org [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Kevin Custer
> > Sent: Wednesday, September 11, 2013 12:45 PM
> > To: Andreas Pleschutznig
> > Cc: app_rpt-users at ohnosec.org
> > Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar
> >  
> > On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:
> >  
> >  
> > Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time.
> > 
> > Sounds to me like a computer with too little RAM or too little CPU horsepower.
> > 
> > When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.
> > 
> > Kevin
> > 
> > 
> > -- 
> > This message was scanned and is believed to be clean. 
> > Click here to report this message as spam.
> 
> 



From ars.w5omr at gmail.com  Wed Sep 11 23:21:46 2013
From: ars.w5omr at gmail.com (Geoff Edmonson)
Date: Wed, 11 Sep 2013 18:21:46 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
Message-ID: <bv3qcwew8jccbv0n95dv3xaa.1378941706294@email.android.com>

Which channel driver (chan_?) are you using?

From ke6pcv at cal-net.org  Wed Sep 11 23:41:34 2013
From: ke6pcv at cal-net.org (Marshall Oldham)
Date: Wed, 11 Sep 2013 16:41:34 -0700
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <F7CA14B6-B9C0-44C9-AF53-48FA72A9CA95@repeater.net>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org><52305933.2040001@cogeco.ca><80B34670-B925-4472-9692-60B46A306820@ple.org><52309E27.5080209@kuggie.com><4BCC91CBCFD66C4489B4BD3233140C3E048596B6D2C2@exchange.mail.repeater.net><B6385102-C597-4035-921D-DD29339DF61E@ple.org>
	<F7CA14B6-B9C0-44C9-AF53-48FA72A9CA95@repeater.net>
Message-ID: <75FE677DBE9048E4AE1558F42722FBBB@marshall2>

I had sound related problems with a AMD processor as well many years ago. Finally gave up and used an Intel based and all the sound
problems went away.
 
Marshall - ke6pcv 



  _____  

From: app_rpt-users-bounces at ohnosec.org [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Corey Dean
Sent: Wednesday, September 11, 2013 4:03 PM
To: Andreas Pleschutznig
Cc: app_rpt-users at ohnosec.org
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar


I have nothing but problems with an amd processor that I was using as my first node.  A lot of it was audio related as well.  I
can't remember if it was Steve or Jim that ask me the very same question.  I ended up trying an intel CPU on a pretty Low end
machine and all my problems went away.  I have built many nodes since for both myself and others and have never had that problem
show up again.

Corey n3fe
Sent from my iPhone

On Sep 11, 2013, at 6:39 PM, "Andreas Pleschutznig" <andy at ple.org> wrote:



Huh? Now you see me stumped. I can see some badly designed math library having a problem with that, but what does that have to do
with the processor. I am not discounting the statement, just trying to gather information, and the statement does not just come
through the filter I have in my brain as to makes sense. 

As far as the CPU goes, yes I do have an AMD chip:

[root at s02 proc]# cat /proc/cpuinfo
processor : 0
vendor_id : AuthenticAMD
cpu family : 20
model : 1
model name : AMD E-350 Processor
stepping : 0
cpu MHz : 1597.148
cache size : 512 KB
physical id : 0
siblings : 2
core id : 0
cpu cores : 2
apicid : 0
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 6
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext
fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a
misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
bogomips : 3194.29

processor : 1
vendor_id : AuthenticAMD
cpu family : 20
model : 1
model name : AMD E-350 Processor
stepping : 0
cpu MHz : 1597.148
cache size : 512 KB
physical id : 0
siblings : 2
core id : 1
cpu cores : 2
apicid : 1
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 6
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext
fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a
misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
bogomips : 3194.88

[root at s02 proc]# 

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 4:01 PM, Corey Dean <n3fe at repeater.net> wrote:


Any chance you are using an AMD processor?  If you tx tone and your voice are more than 999 combined, that could also be the
problem.

 

Corey N3FE

 

From: app_rpt-users-bounces at ohnosec.org [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Kevin Custer
Sent: Wednesday, September 11, 2013 12:45 PM
To: Andreas Pleschutznig
Cc: app_rpt-users at ohnosec.org
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar



On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:





Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it
does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect
99% of the time. 


Sounds to me like a computer with too little RAM or too little CPU horsepower.

When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.

Kevin


-- 
This message was scanned and is believed to be clean. 
Click  <http://simba.repeater.net:8080/cgi-bin/learn-msg.cgi?id=A3CC12223.A96FE> here to report this message as spam. 


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From andy at ple.org  Wed Sep 11 23:46:54 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Wed, 11 Sep 2013 18:46:54 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <bv3qcwew8jccbv0n95dv3xaa.1378941706294@email.android.com>
References: <bv3qcwew8jccbv0n95dv3xaa.1378941706294@email.android.com>
Message-ID: <84B66F14-27F5-48F4-8BDF-991CF6F46527@ple.org>

Good question. How do I find out?

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 6:21 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:

> Which channel driver (chan_?) are you using?

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From n8ohu at yahoo.com  Thu Sep 12 00:05:03 2013
From: n8ohu at yahoo.com (Matthew Pitts)
Date: Wed, 11 Sep 2013 17:05:03 -0700 (PDT)
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <Pine.LNX.4.44.1309111838300.15898-100000@goliath.inttek.net>
References: <B6385102-C597-4035-921D-DD29339DF61E@ple.org>
	<Pine.LNX.4.44.1309111838300.15898-100000@goliath.inttek.net>
Message-ID: <1378944303.25850.YahooMailNeo@web141006.mail.bf1.yahoo.com>

Yes, and the problems that occur are usually related to the USB subsystem, based on things I've heard about with other software I run.

Matthew Pitts
N8OHU



________________________________
 From: David McGough <kb4fxc at inttek.net>
To: Andreas Pleschutznig <andy at ple.org> 
Cc: "app_rpt-users at ohnosec.org" <app_rpt-users at ohnosec.org> 
Sent: Wednesday, September 11, 2013 7:06 PM
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar
 


It never fails to amaze me how AMD CPU's get blamed for various
issues--and the actual CPU's are almost never at fault! And, yes, AMD and
Intel have both had an occasional "oops" impacting a specific CPU series.

In my experience, most of these "blame" issues arise from really 
poorly designed/buggy motherboards. Unfortunately, in practice, more of 
these bad boards seem to be found in low-end AMD boxes, where system 
builders are cutting every cost corner.

With a good quality MB, AMD and Intel CPU's both work great and exhibit 
similar reliability/stability....And, in case you're wondering, I run a 
Co-Lo center and have a few racks full of AMD boards! I measure up-times 
in years.

73, David KB4FXC


On Wed, 11 Sep 2013, Andreas Pleschutznig wrote:

> Huh? Now you see me stumped. I can see some badly designed math library 
having a problem with that, but what does that have to do with the 
processor. I am not discounting the statement, just trying to gather 
information, and the statement does not just come through the filter I 
have in my brain as to makes sense.
> 
> As far as the CPU goes, yes I do have an AMD chip:
> 
> [root at s02 proc]# cat /proc/cpuinfo
> processor??? : 0
> vendor_id??? : AuthenticAMD
> cpu family??? : 20
> model??? ??? : 1
> model name??? : AMD E-350 Processor
> stepping??? : 0
> cpu MHz??? ??? : 1597.148
> cache size??? : 512 KB
> physical id??? : 0
> siblings??? : 2
> core id??? ??? : 0
> cpu cores??? : 2
> apicid??? ??? : 0
> fdiv_bug??? : no
> hlt_bug??? ??? : no
> f00f_bug??? : no
> coma_bug??? : no
> fpu??? ??? : yes
> fpu_exception??? : yes
> cpuid level??? : 6
> wp??? ??? : yes
> flags??? ??? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
> bogomips??? : 3194.29
> 
> processor??? : 1
> vendor_id??? : AuthenticAMD
> cpu family??? : 20
> model??? ??? : 1
> model name??? : AMD E-350 Processor
> stepping??? : 0
> cpu MHz??? ??? : 1597.148
> cache size??? : 512 KB
> physical id??? : 0
> siblings??? : 2
> core id??? ??? : 1
> cpu cores??? : 2
> apicid??? ??? : 1
> fdiv_bug??? : no
> hlt_bug??? ??? : no
> f00f_bug??? : no
> coma_bug??? : no
> fpu??? ??? : yes
> fpu_exception??? : yes
> cpuid level??? : 6
> wp??? ??? : yes
> flags??? ??? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
> bogomips??? : 3194.88
> 
> [root at s02 proc]# 
> 
> --
> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>? KA5PLE, Allstar: 29841, Echolonk: 884823
> 
> On Sep 11, 2013, at 4:01 PM, Corey Dean <n3fe at repeater.net> wrote:
> 
> > Any chance you are using an AMD processor?? If you tx tone and your voice are more than 999 combined, that could also be the problem.
> >? 
> > Corey N3FE
> >? 
> > From: app_rpt-users-bounces at ohnosec.org [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Kevin Custer
> > Sent: Wednesday, September 11, 2013 12:45 PM
> > To: Andreas Pleschutznig
> > Cc: app_rpt-users at ohnosec.org
> > Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar
> >? 
> > On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:
> >? 
> >? 
> > Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time.
> > 
> > Sounds to me like a computer with too little RAM or too little CPU horsepower.
> > 
> > When you run the top command, is it using swap memory?? If so, that's bad...and you need more RAM.
> > 
> > Kevin
> > 
> > 
> > -- 
> > This message was scanned and is believed to be clean. 
> > Click here to report this message as spam.
> 
> 

_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From ars.w5omr at gmail.com  Thu Sep 12 00:22:00 2013
From: ars.w5omr at gmail.com (Geoff Edmonson)
Date: Wed, 11 Sep 2013 19:22:00 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
Message-ID: <gjidq59b1o4qhlvnanx5jb53.1378943740227@email.android.com>

In rpt. conf, somewhere near the top just under your node number.  

Simpleusb doesn't utilize as many resources, its my understanding, as usbradio.  

-Geoff/w5omr

Andreas Pleschutznig <andy at ple.org> wrote:

>Good question. How do I find out?
>
>--
>Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
> KA5PLE, Allstar: 29841, Echolonk: 884823
>
>On Sep 11, 2013, at 6:21 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:
>
>> Which channel driver (chan_?) are you using?
>

From andy at ple.org  Thu Sep 12 00:49:46 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Wed, 11 Sep 2013 19:49:46 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <gjidq59b1o4qhlvnanx5jb53.1378943740227@email.android.com>
References: <gjidq59b1o4qhlvnanx5jb53.1378943740227@email.android.com>
Message-ID: <A6436C9B-D363-417E-8E1B-2C2F28CAAF6D@ple.org>

/etc/asterisk/rpt.conf
?
; Radio Repeater configuration file (for use with app_rpt)
;

[29886]                                 ; Change this to your assigned node number

rxchannel=Radio/usb29886
duplex=2
erxgain=-3
etxgain=3
;controlstates=controlstates
scheduler=schedule29886
morse=morse29886
macro=macro29886
functions=functions
phone_functions=functions
link_functions=functions
telemetry=telemetry
wait_times=wait-times
context =  radio
callerid = "Repeater" <0000000000>
idrecording = |iWQRM312
accountcode=RADIO
hangtime=0                              ; squelch tail hang time (in ms) (optional)
althangtime=0                           ; alternate squelch tail
totime=170000
idtime=0
politeid=3000

idtalkover=|iWQRM312
unlinkedct=ct1                          ; unlinked courtesy tone (optional) default is none
;remotect=ct3
linkunkeyct=ct1                        ; courtesy tone sent on link unkey
nolocallinkct=0                         ; Send unlinkedct instead of linkedct if another local node is connected to this node (hosted on the same PC)
nounkeyct=0                             ; Set to a 1 to eliminate courtesy tones and associated delays.
;eannmode=1                             ; Default: 1 = Say only node number on echolink connects
                                        ; 2 = say phonetic call sign only on echolink connects
                                        ; 3 = say phonetic call sign and node number on echolink connects
;connpgm=yourconnectprogram             ; Default: Disabled. Execute a program you specify on connect.
                                        ; passes 2 command line arguments to your program:
                                        ; 1. node number in this stanza (us)
                                        ; 2. node number being connected to us (them)
;discpgm=yourdisconnectprogram          ; Default: Disabled. Execute a program you specify on disconnect.
                                        ; passes 2 command line arguments to your program:
                                        ; 1. node number in this stanza (us)
                                        ; 2. node number being disconnected from us (them)
;lnkactenable=0                         ; Set to 1 to enable the link activity timer. Applicable to standard nodes only.
;lnkacttime=1800                        ; Link activity timer time in seconds.
;lnkactmacro=*52                        ; Function to execute when link activity timer expires.
;lnkacttimerwarn=30seconds              ; Message to play when the link activity timer has 30 seconds left.
;remote_inact_timeout=1800              ; Inactivity timer for remote base nodes only (set to 0 to disable).
;remote_timeout=3600                    ; Session time out for remote base. (set to 0 to disable)
;holdofftelem=0
beaconing=0
?



That what you asked?

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 7:22 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:

> In rpt. conf, somewhere near the top just under your node number.  
> 
> Simpleusb doesn't utilize as many resources, its my understanding, as usbradio.  
> 
> -Geoff/w5omr
> 
> Andreas Pleschutznig <andy at ple.org> wrote:
> 
>> Good question. How do I find out?
>> 
>> --
>> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>> KA5PLE, Allstar: 29841, Echolonk: 884823
>> 
>> On Sep 11, 2013, at 6:21 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:
>> 
>>> Which channel driver (chan_?) are you using?
>> 

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From kuggie at kuggie.com  Thu Sep 12 01:41:00 2013
From: kuggie at kuggie.com (Kevin Custer)
Date: Wed, 11 Sep 2013 21:41:00 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <26C787A0-3D15-4D71-9AD1-0C125EFB52B3@ple.org>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<A94F9FC9-3489-4935-8458-732201FF603E@ple.org>
	<5230D720.1090802@kuggie.com>
	<26C787A0-3D15-4D71-9AD1-0C125EFB52B3@ple.org>
Message-ID: <52311BAC.7050007@kuggie.com>

On 9/11/2013 6:26 PM, Andreas Pleschutznig wrote:
> Hi Kevin, the internet connection if anything is too fast as well. 
> Besides, even if I had no internet connection that should noyt have 
> any impact on time announcements, right?

Correct - I didn't see your earlier post mentioning local telemetry also 
being affected.  Makes me believe, even more, it's a hardware issue.
Do you have another/different computer you can load up?

Kevin
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From n3fe at repeater.net  Thu Sep 12 02:03:05 2013
From: n3fe at repeater.net (Corey Dean)
Date: Wed, 11 Sep 2013 22:03:05 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <Pine.LNX.4.44.1309111838300.15898-100000@goliath.inttek.net>
References: <B6385102-C597-4035-921D-DD29339DF61E@ple.org>
	<Pine.LNX.4.44.1309111838300.15898-100000@goliath.inttek.net>
Message-ID: <4BCC91CBCFD66C4489B4BD3233140C3E048596B6D2C3@exchange.mail.repeater.net>

I only mentioned it because it was mentioned to me from one of the people that helped work on the code.  I wasn't try to start a war against the two giants.  With that said, I will sit back and watch the thread...

Corey n3fe 

-----Original Message-----
From: app_rpt-users-bounces at ohnosec.org [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of David McGough
Sent: Wednesday, September 11, 2013 7:07 PM
To: Andreas Pleschutznig
Cc: app_rpt-users at ohnosec.org
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar


It never fails to amaze me how AMD CPU's get blamed for various issues--and the actual CPU's are almost never at fault! And, yes, AMD and Intel have both had an occasional "oops" impacting a specific CPU series.

In my experience, most of these "blame" issues arise from really poorly designed/buggy motherboards. Unfortunately, in practice, more of these bad boards seem to be found in low-end AMD boxes, where system builders are cutting every cost corner.

With a good quality MB, AMD and Intel CPU's both work great and exhibit similar reliability/stability....And, in case you're wondering, I run a Co-Lo center and have a few racks full of AMD boards! I measure up-times in years.

73, David KB4FXC


On Wed, 11 Sep 2013, Andreas Pleschutznig wrote:

> Huh? Now you see me stumped. I can see some badly designed math 
> library
having a problem with that, but what does that have to do with the processor. I am not discounting the statement, just trying to gather information, and the statement does not just come through the filter I have in my brain as to makes sense.
> 
> As far as the CPU goes, yes I do have an AMD chip:
> 
> [root at s02 proc]# cat /proc/cpuinfo
> processor	: 0
> vendor_id	: AuthenticAMD
> cpu family	: 20
> model		: 1
> model name	: AMD E-350 Processor
> stepping	: 0
> cpu MHz		: 1597.148
> cache size	: 512 KB
> physical id	: 0
> siblings	: 2
> core id		: 0
> cpu cores	: 2
> apicid		: 0
> fdiv_bug	: no
> hlt_bug		: no
> f00f_bug	: no
> coma_bug	: no
> fpu		: yes
> fpu_exception	: yes
> cpuid level	: 6
> wp		: yes
> flags		: fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
> bogomips	: 3194.29
> 
> processor	: 1
> vendor_id	: AuthenticAMD
> cpu family	: 20
> model		: 1
> model name	: AMD E-350 Processor
> stepping	: 0
> cpu MHz		: 1597.148
> cache size	: 512 KB
> physical id	: 0
> siblings	: 2
> core id		: 1
> cpu cores	: 2
> apicid		: 1
> fdiv_bug	: no
> hlt_bug		: no
> f00f_bug	: no
> coma_bug	: no
> fpu		: yes
> fpu_exception	: yes
> cpuid level	: 6
> wp		: yes
> flags		: fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt pdpe1gb rdtscp lm constant_tsc nonstop_tsc pni ssse3 cx16 popcnt lahf_lm cmp_legacy svm extapic cr8legacy abm sse4a misalignsse 3dnowprefetch ibs skinit wdt ts ttp tm stc 100mhzsteps hwpstate [8]
> bogomips	: 3194.88
> 
> [root at s02 proc]#
> 
> --
> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook  KA5PLE, 
> Allstar: 29841, Echolonk: 884823
> 
> On Sep 11, 2013, at 4:01 PM, Corey Dean <n3fe at repeater.net> wrote:
> 
> > Any chance you are using an AMD processor?  If you tx tone and your voice are more than 999 combined, that could also be the problem.
> >  
> > Corey N3FE
> >  
> > From: app_rpt-users-bounces at ohnosec.org 
> > [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Kevin Custer
> > Sent: Wednesday, September 11, 2013 12:45 PM
> > To: Andreas Pleschutznig
> > Cc: app_rpt-users at ohnosec.org
> > Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar
> >  
> > On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:
> >  
> >  
> > Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time.
> > 
> > Sounds to me like a computer with too little RAM or too little CPU horsepower.
> > 
> > When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.
> > 
> > Kevin
> > 
> > 
> > --
> > This message was scanned and is believed to be clean. 
> > Click here to report this message as spam.
> 
> 

_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

--
This message was scanned and is believed to be clean.
Click here to report this message as spam. 
http://simba.repeater.net/cgi-bin/learn-msg.cgi?id=23C952223.A8B7C




From n3ssl at yahoo.com  Thu Sep 12 02:07:54 2013
From: n3ssl at yahoo.com (Ryan Gross)
Date: Wed, 11 Sep 2013 19:07:54 -0700 (PDT)
Subject: [App_rpt-users] Distorted audio
Message-ID: <1378951674.30020.YahooMailNeo@web141506.mail.bf1.yahoo.com>

Ran into this problem a while ago and was found the URI interface was the problem with similar symptoms. Is this happening in duplex or simplex with audio ?
?
Ryan n3ssl 
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From george at dyb.com  Thu Sep 12 03:41:53 2013
From: george at dyb.com (George Csahanin)
Date: Wed, 11 Sep 2013 22:41:53 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<657AC31609934FC08D547E3EF8FFBEFA@georgecPCOMVC>
	<605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>
Message-ID: <D70210BB8E784E08BD909687B264FC1C@georgecPCOMVC>

10-4, figured that out long time ago, but all the interest in using DSP in this and I thought it would work. It might on a different platform. I opened a Mantis ticket on it a few years ago. But it really is a reason to stick with PL decode and squelch as done by Motorola and GE, why re-invent the wheel? I have one remote ?private? node using a HT10 and DSP. Works fine for what it is. 

Georgec

From: Bryan D. Boyle 
Sent: Wednesday, September 11, 2013 12:56 PM
To: George Csahanin 
Cc: Kevin Custer ; Andreas Pleschutznig ; app_rpt-users at ohnosec.org 
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar

DSP takes a significant amount of horsepower.  Ya don't get something for nothing; reading a hardware status bit or two is a lot less intensive at the expense of additional wiring and configuration statements in the config files than decoding the stream, deriving the signals, etc.  any overhead in doing so is bound to have an effect on other streams going on at that time.

and we humans don't process breaks in audio as we do with visual perceptions.

-- 
Bryan
Sent from my iPhone 5...small
keyboard, big fingers...please
forgive misspellings...



On Sep 11, 2013, at 13:44, "George Csahanin" <george at dyb.com> wrote:


  I have that exact problem if I use DSP, moved to COS and PTT on DB25 and it stops. Did this in every case I tried, though the only platform I used was a D945 or D201 Intel board.

  Ran 1GB of RAM and CPU is whatever a D945 is. 

  GeorgeC
  W2DB
  2360


  From: Kevin Custer 
  Sent: Wednesday, September 11, 2013 11:45 AM
  To: Andreas Pleschutznig 
  Cc: app_rpt-users at ohnosec.org 
  Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar

  On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:



    Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time. 


  Sounds to me like a computer with too little RAM or too little CPU horsepower.

  When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.

  Kevin


  _______________________________________________
  App_rpt-users mailing list
  App_rpt-users at ohnosec.org
  http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From george at dyb.com  Thu Sep 12 03:43:31 2013
From: george at dyb.com (George Csahanin)
Date: Wed, 11 Sep 2013 22:43:31 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <4FB428D6-B4D1-4CA6-8528-8C9B8CF01844@ple.org>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<657AC31609934FC08D547E3EF8FFBEFA@georgecPCOMVC>
	<605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>
	<4FB428D6-B4D1-4CA6-8528-8C9B8CF01844@ple.org>
Message-ID: <F58664340100436CA409CCAE4AD4E6B4@georgecPCOMVC>

Mine would do it  on local as well as remote and repeat. Heck, even radio tune txaudio tone had it, pl tone had it, etc.

G

From: Andreas Pleschutznig 
Sent: Wednesday, September 11, 2013 1:06 PM
To: Bryan D. Boyle 
Cc: George Csahanin ; Kevin Custer ; app_rpt-users at ohnosec.org 
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar

George, Bryan, 

the problem *I* have is not limited to a repeat. it does happen when on simple time announcements. At the time when it happens the system is at less than 35% CPU, so I kind of doubt this is a problem in the CPU being too slow. 

--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 12:56 PM, Bryan D. Boyle <bdboyle at bdboyle.com> wrote:


  DSP takes a significant amount of horsepower.  Ya don't get something for nothing; reading a hardware status bit or two is a lot less intensive at the expense of additional wiring and configuration statements in the config files than decoding the stream, deriving the signals, etc.  any overhead in doing so is bound to have an effect on other streams going on at that time.

  and we humans don't process breaks in audio as we do with visual perceptions.

  -- 
  Bryan
  Sent from my iPhone 5...small
  keyboard, big fingers...please
  forgive misspellings...



  On Sep 11, 2013, at 13:44, "George Csahanin" <george at dyb.com> wrote:


    I have that exact problem if I use DSP, moved to COS and PTT on DB25 and it stops. Did this in every case I tried, though the only platform I used was a D945 or D201 Intel board.

    Ran 1GB of RAM and CPU is whatever a D945 is. 

    GeorgeC
    W2DB
    2360


    From: Kevin Custer 
    Sent: Wednesday, September 11, 2013 11:45 AM
    To: Andreas Pleschutznig 
    Cc: app_rpt-users at ohnosec.org 
    Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar

    On 9/11/2013 12:20 PM, Andreas Pleschutznig wrote:



      Best I can describe the effect is that the audio is time sliced in 20ms packets and every other packet is missing, left out. and it does not happen when audio is especially loud or quiet. Sometimes it happens on the time announcements, which are absolutely perfect 99% of the time. 


    Sounds to me like a computer with too little RAM or too little CPU horsepower.

    When you run the top command, is it using swap memory?  If so, that's bad...and you need more RAM.

    Kevin


    _______________________________________________
    App_rpt-users mailing list
    App_rpt-users at ohnosec.org
    http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From wb3awj at comcast.net  Thu Sep 12 04:12:50 2013
From: wb3awj at comcast.net (Robert A. Poff)
Date: Thu, 12 Sep 2013 00:12:50 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <D70210BB8E784E08BD909687B264FC1C@georgecPCOMVC>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<657AC31609934FC08D547E3EF8FFBEFA@georgecPCOMVC>
	<605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>
	<D70210BB8E784E08BD909687B264FC1C@georgecPCOMVC>
Message-ID: <9e933441-86e0-4307-82d7-e50029d0ea56@email.android.com>

Meanwhile on my nodes, here we're running full DSP with URIs  on 3 GHz P4's with 2GB (IBM Think Center SFF) of memory with no problems. 

-- 
Sent from my Android phone with K-9 Mail. Please excuse my brevity and typing errors.
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From tim.sawyer at me.com  Thu Sep 12 04:37:29 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Wed, 11 Sep 2013 21:37:29 -0700
Subject: [App_rpt-users] Lots of questions
In-Reply-To: <CAM09xEaBYxEgUR96N6JMN8-MGs-zw-vcmimdP-ki=xgdRxw=XA@mail.gmail.com>
References: <CAM09xEbTCCp9LfwqiFz2xU=Y=Sg+iZFAhukfMaLyVpz5AuKhHQ@mail.gmail.com>
	<CAM09xEaBYxEgUR96N6JMN8-MGs-zw-vcmimdP-ki=xgdRxw=XA@mail.gmail.com>
Message-ID: <132AFE93-8CC9-4B20-8A92-CE0093736A58@me.com>

app_rpt has echolink built into it. See http://ohnosec.org/drupal/node/56
--
Tim
:wq

On Sep 11, 2013, at 12:14 PM, Robert Newberry <N1XBM at amsat.org> wrote:

> I have a mastr ii repeater. I'd like to use app-rpt as the controller for the repeater. I'd like to be able to run echolink off the repeater. I also have a URI interface. I'd also like to after one repeater is on the air and available on echolink be able to do the same thing to another repeater to make a linked network via IP
> 
> My questions are can echolink and app_rpt be run at the same time? At the same time providing repeater control and echolink service.
> 
> Has anyone had any success using a raspberry pi with app_rpt and a URI? I have a laptop I could use, but the pi is an attractive solution.
> 
> I'd also like to add I'm new to linux. I'm OK on computers and networking I also work in land mobile radio so the RF end of it I'm good.
> 
> Apparare Scientior 
> Paratus Communicare
> -N1XBM
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From tim.sawyer at me.com  Thu Sep 12 04:48:38 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Wed, 11 Sep 2013 21:48:38 -0700
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <A6436C9B-D363-417E-8E1B-2C2F28CAAF6D@ple.org>
References: <gjidq59b1o4qhlvnanx5jb53.1378943740227@email.android.com>
	<A6436C9B-D363-417E-8E1B-2C2F28CAAF6D@ple.org>
Message-ID: <ABEF8C12-D064-413D-BF28-D34FAB5F154A@me.com>

That's USBradio which has the DSP stuff in it. If you can get COR and CTCSS detect out of your radio you are better off switching to simpleUSB.
See http://ohnosec.org/drupal/node/193
--
Tim
:wq

On Sep 11, 2013, at 5:49 PM, Andreas Pleschutznig <andy at ple.org> wrote:

> /etc/asterisk/rpt.conf
> ?
> ; Radio Repeater configuration file (for use with app_rpt)
> ;
> 
> [29886]                                 ; Change this to your assigned node number
> 
> rxchannel=Radio/usb29886
> duplex=2
> erxgain=-3
> etxgain=3
> ;controlstates=controlstates
> scheduler=schedule29886
> morse=morse29886
> macro=macro29886
> functions=functions
> phone_functions=functions
> link_functions=functions
> telemetry=telemetry
> wait_times=wait-times
> context =  radio
> callerid = "Repeater" <0000000000>
> idrecording = |iWQRM312
> accountcode=RADIO
> hangtime=0                              ; squelch tail hang time (in ms) (optional)
> althangtime=0                           ; alternate squelch tail
> totime=170000
> idtime=0
> politeid=3000
> 
> idtalkover=|iWQRM312
> unlinkedct=ct1                          ; unlinked courtesy tone (optional) default is none
> ;remotect=ct3
> linkunkeyct=ct1                        ; courtesy tone sent on link unkey
> nolocallinkct=0                         ; Send unlinkedct instead of linkedct if another local node is connected to this node (hosted on the same PC)
> nounkeyct=0                             ; Set to a 1 to eliminate courtesy tones and associated delays.
> ;eannmode=1                             ; Default: 1 = Say only node number on echolink connects
>                                       ; 2 = say phonetic call sign only on echolink connects
>                                       ; 3 = say phonetic call sign and node number on echolink connects
> ;connpgm=yourconnectprogram             ; Default: Disabled. Execute a program you specify on connect.
>                                       ; passes 2 command line arguments to your program:
>                                       ; 1. node number in this stanza (us)
>                                       ; 2. node number being connected to us (them)
> ;discpgm=yourdisconnectprogram          ; Default: Disabled. Execute a program you specify on disconnect.
>                                       ; passes 2 command line arguments to your program:
>                                       ; 1. node number in this stanza (us)
>                                       ; 2. node number being disconnected from us (them)
> ;lnkactenable=0                         ; Set to 1 to enable the link activity timer. Applicable to standard nodes only.
> ;lnkacttime=1800                        ; Link activity timer time in seconds.
> ;lnkactmacro=*52                        ; Function to execute when link activity timer expires.
> ;lnkacttimerwarn=30seconds              ; Message to play when the link activity timer has 30 seconds left.
> ;remote_inact_timeout=1800              ; Inactivity timer for remote base nodes only (set to 0 to disable).
> ;remote_timeout=3600                    ; Session time out for remote base. (set to 0 to disable)
> ;holdofftelem=0
> beaconing=0
> ?
> 
> 
> 
> That what you asked?
> 
> --
> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
> KA5PLE, Allstar: 29841, Echolonk: 884823
> 
> On Sep 11, 2013, at 7:22 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:
> 
>> In rpt. conf, somewhere near the top just under your node number.  
>> 
>> Simpleusb doesn't utilize as many resources, its my understanding, as usbradio.  
>> 
>> -Geoff/w5omr
>> 
>> Andreas Pleschutznig <andy at ple.org> wrote:
>> 
>>> Good question. How do I find out?
>>> 
>>> --
>>> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>>> KA5PLE, Allstar: 29841, Echolonk: 884823
>>> 
>>> On Sep 11, 2013, at 6:21 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:
>>> 
>>>> Which channel driver (chan_?) are you using?
>>> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



From bdboyle at bdboyle.com  Thu Sep 12 12:35:14 2013
From: bdboyle at bdboyle.com (Bryan D. Boyle)
Date: Thu, 12 Sep 2013 08:35:14 -0400
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <9e933441-86e0-4307-82d7-e50029d0ea56@email.android.com>
References: <5A70B37C-F9F6-4F52-9CCD-0059D3CABA4D@ple.org>
	<52305933.2040001@cogeco.ca>
	<80B34670-B925-4472-9692-60B46A306820@ple.org>
	<52309E27.5080209@kuggie.com>
	<657AC31609934FC08D547E3EF8FFBEFA@georgecPCOMVC>
	<605BDB62-97C3-4A32-A023-EEFB9E3EEBFB@bdboyle.com>
	<D70210BB8E784E08BD909687B264FC1C@georgecPCOMVC>
	<9e933441-86e0-4307-82d7-e50029d0ea56@email.android.com>
Message-ID: <C2731D35-CFD1-47F0-B4D8-DC8B8FA45606@bdboyle.com>

it is more a function, i think, and this is based on my experiments, of how well the USB was executed on the motherboard, or whether it was done just 'good enough' to handle the predicted keyboard and mouse and probably not much else.  

i had the same problems when i migrated my 3 nodes off a toshiba laptop (!) onto an HP mini-tower, strangely enough.  ran fine on the laptop.  stuttering and drop outs on the HP, which was 4 years newer.  

put in a USB card on the bus rather than using the motherboard USB ports.  has been working fine for the last 2 years.  has to do with the priorities on the motherboard, speed at which the packets can be processed, bios to handle the device routines, etc.  Add in the psychoacoustic effects...it's not as clear cut as to why or where the issue lies.  

but, consider that USB is a compromise design...you want real time industrial strength audio and control, i think Duuude may have some PRI boards still available which will do the trick...:)

--
Bryan
Sent from my iPhone 5...small
keyboard, big fingers...please
forgive misspellings...



On Sep 12, 2013, at 0:12, "Robert A. Poff" <wb3awj at comcast.net> wrote:

> Meanwhile on my nodes, here we're running full DSP with URIs on 3 GHz P4's with 2GB (IBM Think Center SFF) of memory with no problems. 
> 
> -- 
> Sent from my Android phone with K-9 Mail. Please excuse my brevity and typing errors.
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users


From george at dyb.com  Thu Sep 12 14:25:41 2013
From: george at dyb.com (George Csahanin)
Date: Thu, 12 Sep 2013 09:25:41 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <ABEF8C12-D064-413D-BF28-D34FAB5F154A@me.com>
References: <gjidq59b1o4qhlvnanx5jb53.1378943740227@email.android.com><A6436C9B-D363-417E-8E1B-2C2F28CAAF6D@ple.org>
	<ABEF8C12-D064-413D-BF28-D34FAB5F154A@me.com>
Message-ID: <19C85F1D60D549A3BF9778B72353019A@georgecPCOMVC>

just a thought. Maybe a line in drupal node 193 that warns that depending on 
the PC's USB handling the DSP may or may not work well. I know originally I 
figured that the "word' was 'better than 1200 mhz' CPU so the D201 and D945 
boards would be fine, but because of this USB issue it isn't, not a big deal 
because I just extracted COS and PTT from the radio.

But I will say that some of it may be the kernel's handling of things. Now 
I'm way above my pay grade here, but I did connect same USB fob to the same 
PC running Winders XP. Played steady tone out and it was fine and pure with 
no interruptions. Boot back to ACID (Limey does it too) and play out a tone 
(radio tune txvoice) and it had the interruptions. Maybe modifying the 
kernel options? I recall there being different options for various chipsets 
and so on, and USB is in there too...

Back to my cave in Jersey...

GeorgeC
W2DB



-----Original Message----- 
From: Tim Sawyer
Sent: Wednesday, September 11, 2013 11:48 PM
To: app_rpt-users at ohnosec.org list
Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar

That's USBradio which has the DSP stuff in it. If you can get COR and CTCSS 
detect out of your radio you are better off switching to simpleUSB.
See http://ohnosec.org/drupal/node/193
--
Tim
:wq

On Sep 11, 2013, at 5:49 PM, Andreas Pleschutznig <andy at ple.org> wrote:

> /etc/asterisk/rpt.conf
> ?
> ; Radio Repeater configuration file (for use with app_rpt)
> ;
>
> [29886]                                 ; Change this to your assigned 
> node number
>
> rxchannel=Radio/usb29886
> duplex=2
> erxgain=-3
> etxgain=3
> ;controlstates=controlstates
> scheduler=schedule29886
> morse=morse29886
> macro=macro29886
> functions=functions
> phone_functions=functions
> link_functions=functions
> telemetry=telemetry
> wait_times=wait-times
> context =  radio
> callerid = "Repeater" <0000000000>
> idrecording = |iWQRM312
> accountcode=RADIO
> hangtime=0                              ; squelch tail hang time (in ms) 
> (optional)
> althangtime=0                           ; alternate squelch tail
> totime=170000
> idtime=0
> politeid=3000
>
> idtalkover=|iWQRM312
> unlinkedct=ct1                          ; unlinked courtesy tone 
> (optional) default is none
> ;remotect=ct3
> linkunkeyct=ct1                        ; courtesy tone sent on link unkey
> nolocallinkct=0                         ; Send unlinkedct instead of 
> linkedct if another local node is connected to this node (hosted on the 
> same PC)
> nounkeyct=0                             ; Set to a 1 to eliminate courtesy 
> tones and associated delays.
> ;eannmode=1                             ; Default: 1 = Say only node 
> number on echolink connects
>                                       ; 2 = say phonetic call sign only on 
> echolink connects
>                                       ; 3 = say phonetic call sign and 
> node number on echolink connects
> ;connpgm=yourconnectprogram             ; Default: Disabled. Execute a 
> program you specify on connect.
>                                       ; passes 2 command line arguments to 
> your program:
>                                       ; 1. node number in this stanza (us)
>                                       ; 2. node number being connected to 
> us (them)
> ;discpgm=yourdisconnectprogram          ; Default: Disabled. Execute a 
> program you specify on disconnect.
>                                       ; passes 2 command line arguments to 
> your program:
>                                       ; 1. node number in this stanza (us)
>                                       ; 2. node number being disconnected 
> from us (them)
> ;lnkactenable=0                         ; Set to 1 to enable the link 
> activity timer. Applicable to standard nodes only.
> ;lnkacttime=1800                        ; Link activity timer time in 
> seconds.
> ;lnkactmacro=*52                        ; Function to execute when link 
> activity timer expires.
> ;lnkacttimerwarn=30seconds              ; Message to play when the link 
> activity timer has 30 seconds left.
> ;remote_inact_timeout=1800              ; Inactivity timer for remote base 
> nodes only (set to 0 to disable).
> ;remote_timeout=3600                    ; Session time out for remote 
> base. (set to 0 to disable)
> ;holdofftelem=0
> beaconing=0
> ?
>
>
>
> That what you asked?
>
> --
> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
> KA5PLE, Allstar: 29841, Echolonk: 884823
>
> On Sep 11, 2013, at 7:22 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:
>
>> In rpt. conf, somewhere near the top just under your node number.
>>
>> Simpleusb doesn't utilize as many resources, its my understanding, as 
>> usbradio.
>>
>> -Geoff/w5omr
>>
>> Andreas Pleschutznig <andy at ple.org> wrote:
>>
>>> Good question. How do I find out?
>>>
>>> --
>>> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>>> KA5PLE, Allstar: 29841, Echolonk: 884823
>>>
>>> On Sep 11, 2013, at 6:21 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:
>>>
>>>> Which channel driver (chan_?) are you using?
>>>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users




From N1XBM at amsat.org  Thu Sep 12 19:50:18 2013
From: N1XBM at amsat.org (Robert Newberry)
Date: Thu, 12 Sep 2013 15:50:18 -0400
Subject: [App_rpt-users] Lots of questions
In-Reply-To: <132AFE93-8CC9-4B20-8A92-CE0093736A58@me.com>
References: <CAM09xEbTCCp9LfwqiFz2xU=Y=Sg+iZFAhukfMaLyVpz5AuKhHQ@mail.gmail.com>
	<CAM09xEaBYxEgUR96N6JMN8-MGs-zw-vcmimdP-ki=xgdRxw=XA@mail.gmail.com>
	<132AFE93-8CC9-4B20-8A92-CE0093736A58@me.com>
Message-ID: <CAM09xEZtmExqUdCzsoYwXvQsH6GT4hZ=_uBymvn0RCR321drAA@mail.gmail.com>

OK so echolink...check!

What about using raspberry pi? I have a laptop I can dedicate, but the pi
looks appealing.

Apparare Scientior
Paratus Communicare
-N1XBM
On Sep 12, 2013 12:37 AM, "Tim Sawyer" <tim.sawyer at me.com> wrote:

> app_rpt has echolink built into it. See http://ohnosec.org/drupal/node/56
> --
> Tim
> :wq
>
> On Sep 11, 2013, at 12:14 PM, Robert Newberry <N1XBM at amsat.org> wrote:
>
> I have a mastr ii repeater. I'd like to use app-rpt as the controller for
> the repeater. I'd like to be able to run echolink off the repeater. I also
> have a URI interface. I'd also like to after one repeater is on the air and
> available on echolink be able to do the same thing to another repeater to
> make a linked network via IP
>
> My questions are can echolink and app_rpt be run at the same time? At the
> same time providing repeater control and echolink service.
>
> Has anyone had any success using a raspberry pi with app_rpt and a URI? I
> have a laptop I could use, but the pi is an attractive solution.
>
> I'd also like to add I'm new to linux. I'm OK on computers and networking
> I also work in land mobile radio so the RF end of it I'm good.
>
> Apparare Scientior
> Paratus Communicare
> -N1XBM
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
>
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From rpt2 at chuck.midlandsnetworking.com  Thu Sep 12 19:53:06 2013
From: rpt2 at chuck.midlandsnetworking.com (Chuck Henderson)
Date: Thu, 12 Sep 2013 14:53:06 -0500
Subject: [App_rpt-users] Distorted audio in app_rpt/Allstar
In-Reply-To: <19C85F1D60D549A3BF9778B72353019A@georgecPCOMVC>
References: <gjidq59b1o4qhlvnanx5jb53.1378943740227@email.android.com>
	<A6436C9B-D363-417E-8E1B-2C2F28CAAF6D@ple.org>
	<ABEF8C12-D064-413D-BF28-D34FAB5F154A@me.com>
	<19C85F1D60D549A3BF9778B72353019A@georgecPCOMVC>
Message-ID: <CADEYVwhM0tPbQYhmOxVVMEy9hDR8Am9cP4Cxr23tFL2GQ6CGoA@mail.gmail.com>

I just noticed this...

>when it does that the system is at less than 35% CPU and still has about 3GB of RAM open (this is on a 4GB system)

so I checked my atom d510mo system and it seems to always be at 1% CPU
or less.  So why is yours running 35%?  You must have something other
than the normal ACID install running to have it that high with a dual
core 2+GHZ cpu.


On Thu, Sep 12, 2013 at 9:25 AM, George Csahanin <george at dyb.com> wrote:
> just a thought. Maybe a line in drupal node 193 that warns that depending on
> the PC's USB handling the DSP may or may not work well. I know originally I
> figured that the "word' was 'better than 1200 mhz' CPU so the D201 and D945
> boards would be fine, but because of this USB issue it isn't, not a big deal
> because I just extracted COS and PTT from the radio.
>
> But I will say that some of it may be the kernel's handling of things. Now
> I'm way above my pay grade here, but I did connect same USB fob to the same
> PC running Winders XP. Played steady tone out and it was fine and pure with
> no interruptions. Boot back to ACID (Limey does it too) and play out a tone
> (radio tune txvoice) and it had the interruptions. Maybe modifying the
> kernel options? I recall there being different options for various chipsets
> and so on, and USB is in there too...
>
> Back to my cave in Jersey...
>
> GeorgeC
> W2DB
>
>
>
> -----Original Message----- From: Tim Sawyer
> Sent: Wednesday, September 11, 2013 11:48 PM
> To: app_rpt-users at ohnosec.org list
>
> Subject: Re: [App_rpt-users] Distorted audio in app_rpt/Allstar
>
> That's USBradio which has the DSP stuff in it. If you can get COR and CTCSS
> detect out of your radio you are better off switching to simpleUSB.
> See http://ohnosec.org/drupal/node/193
> --
> Tim
> :wq
>
> On Sep 11, 2013, at 5:49 PM, Andreas Pleschutznig <andy at ple.org> wrote:
>
>> /etc/asterisk/rpt.conf
>> ?
>> ; Radio Repeater configuration file (for use with app_rpt)
>> ;
>>
>> [29886]                                 ; Change this to your assigned
>> node number
>>
>> rxchannel=Radio/usb29886
>> duplex=2
>> erxgain=-3
>> etxgain=3
>> ;controlstates=controlstates
>> scheduler=schedule29886
>> morse=morse29886
>> macro=macro29886
>> functions=functions
>> phone_functions=functions
>> link_functions=functions
>> telemetry=telemetry
>> wait_times=wait-times
>> context =  radio
>> callerid = "Repeater" <0000000000>
>> idrecording = |iWQRM312
>> accountcode=RADIO
>> hangtime=0                              ; squelch tail hang time (in ms)
>> (optional)
>> althangtime=0                           ; alternate squelch tail
>> totime=170000
>> idtime=0
>> politeid=3000
>>
>> idtalkover=|iWQRM312
>> unlinkedct=ct1                          ; unlinked courtesy tone
>> (optional) default is none
>> ;remotect=ct3
>> linkunkeyct=ct1                        ; courtesy tone sent on link unkey
>> nolocallinkct=0                         ; Send unlinkedct instead of
>> linkedct if another local node is connected to this node (hosted on the same
>> PC)
>> nounkeyct=0                             ; Set to a 1 to eliminate courtesy
>> tones and associated delays.
>> ;eannmode=1                             ; Default: 1 = Say only node
>> number on echolink connects
>>                                       ; 2 = say phonetic call sign only on
>> echolink connects
>>                                       ; 3 = say phonetic call sign and
>> node number on echolink connects
>> ;connpgm=yourconnectprogram             ; Default: Disabled. Execute a
>> program you specify on connect.
>>                                       ; passes 2 command line arguments to
>> your program:
>>                                       ; 1. node number in this stanza (us)
>>                                       ; 2. node number being connected to
>> us (them)
>> ;discpgm=yourdisconnectprogram          ; Default: Disabled. Execute a
>> program you specify on disconnect.
>>                                       ; passes 2 command line arguments to
>> your program:
>>                                       ; 1. node number in this stanza (us)
>>                                       ; 2. node number being disconnected
>> from us (them)
>> ;lnkactenable=0                         ; Set to 1 to enable the link
>> activity timer. Applicable to standard nodes only.
>> ;lnkacttime=1800                        ; Link activity timer time in
>> seconds.
>> ;lnkactmacro=*52                        ; Function to execute when link
>> activity timer expires.
>> ;lnkacttimerwarn=30seconds              ; Message to play when the link
>> activity timer has 30 seconds left.
>> ;remote_inact_timeout=1800              ; Inactivity timer for remote base
>> nodes only (set to 0 to disable).
>> ;remote_timeout=3600                    ; Session time out for remote
>> base. (set to 0 to disable)
>> ;holdofftelem=0
>> beaconing=0
>> ?
>>
>>
>>
>> That what you asked?
>>
>> --
>> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>> KA5PLE, Allstar: 29841, Echolonk: 884823
>>
>> On Sep 11, 2013, at 7:22 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:
>>
>>> In rpt. conf, somewhere near the top just under your node number.
>>>
>>> Simpleusb doesn't utilize as many resources, its my understanding, as
>>> usbradio.
>>>
>>> -Geoff/w5omr
>>>
>>> Andreas Pleschutznig <andy at ple.org> wrote:
>>>
>>>> Good question. How do I find out?
>>>>
>>>> --
>>>> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>>>> KA5PLE, Allstar: 29841, Echolonk: 884823
>>>>
>>>> On Sep 11, 2013, at 6:21 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:
>>>>
>>>>> Which channel driver (chan_?) are you using?
>>>>
>>>>
>>
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users


From bdboyle at bdboyle.com  Thu Sep 12 20:04:57 2013
From: bdboyle at bdboyle.com (Bryan D. Boyle)
Date: Thu, 12 Sep 2013 16:04:57 -0400
Subject: [App_rpt-users] Lots of questions
In-Reply-To: <CAM09xEZtmExqUdCzsoYwXvQsH6GT4hZ=_uBymvn0RCR321drAA@mail.gmail.com>
References: <CAM09xEbTCCp9LfwqiFz2xU=Y=Sg+iZFAhukfMaLyVpz5AuKhHQ@mail.gmail.com>
	<CAM09xEaBYxEgUR96N6JMN8-MGs-zw-vcmimdP-ki=xgdRxw=XA@mail.gmail.com>
	<132AFE93-8CC9-4B20-8A92-CE0093736A58@me.com>
	<CAM09xEZtmExqUdCzsoYwXvQsH6GT4hZ=_uBymvn0RCR321drAA@mail.gmail.com>
Message-ID: <04CD4D41-ECE8-44B6-8303-0A68DAA26C88@bdboyle.com>

popping the popcorn and sitting back...lol.

--
Bryan
Sent from my iPhone 5...small
keyboard, big fingers...please
forgive misspellings...



On Sep 12, 2013, at 15:50, Robert Newberry <N1XBM at amsat.org> wrote:

> OK so echolink...check!
> 
> What about using raspberry pi? I have a laptop I can dedicate, but the pi looks appealing.
> 
> Apparare Scientior 
> Paratus Communicare
> -N1XBM
> 
> On Sep 12, 2013 12:37 AM, "Tim Sawyer" <tim.sawyer at me.com> wrote:
>> app_rpt has echolink built into it. See http://ohnosec.org/drupal/node/56
>> --
>> Tim
>> :wq
>> 
>> On Sep 11, 2013, at 12:14 PM, Robert Newberry <N1XBM at amsat.org> wrote:
>> 
>>> I have a mastr ii repeater. I'd like to use app-rpt as the controller for the repeater. I'd like to be able to run echolink off the repeater. I also have a URI interface. I'd also like to after one repeater is on the air and available on echolink be able to do the same thing to another repeater to make a linked network via IP
>>> 
>>> My questions are can echolink and app_rpt be run at the same time? At the same time providing repeater control and echolink service.
>>> 
>>> Has anyone had any success using a raspberry pi with app_rpt and a URI? I have a laptop I could use, but the pi is an attractive solution.
>>> 
>>> I'd also like to add I'm new to linux. I'm OK on computers and networking I also work in land mobile radio so the RF end of it I'm good.
>>> 
>>> Apparare Scientior 
>>> Paratus Communicare
>>> -N1XBM
>>> 
>>> _______________________________________________
>>> App_rpt-users mailing list
>>> App_rpt-users at ohnosec.org
>>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From tim.sawyer at me.com  Thu Sep 12 21:44:16 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Thu, 12 Sep 2013 14:44:16 -0700
Subject: [App_rpt-users] Lots of questions
In-Reply-To: <CAM09xEZtmExqUdCzsoYwXvQsH6GT4hZ=_uBymvn0RCR321drAA@mail.gmail.com>
References: <CAM09xEbTCCp9LfwqiFz2xU=Y=Sg+iZFAhukfMaLyVpz5AuKhHQ@mail.gmail.com>
	<CAM09xEaBYxEgUR96N6JMN8-MGs-zw-vcmimdP-ki=xgdRxw=XA@mail.gmail.com>
	<132AFE93-8CC9-4B20-8A92-CE0093736A58@me.com>
	<CAM09xEZtmExqUdCzsoYwXvQsH6GT4hZ=_uBymvn0RCR321drAA@mail.gmail.com>
Message-ID: <8D0FE88D-F163-4A3C-85ED-66CB3431BB56@me.com>

In a word, no.
--
Tim
:wq

On Sep 12, 2013, at 12:50 PM, Robert Newberry <N1XBM at amsat.org> wrote:

> What about using raspberry pi?

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From N1XBM at amsat.org  Thu Sep 12 21:59:30 2013
From: N1XBM at amsat.org (Robert Newberry)
Date: Thu, 12 Sep 2013 17:59:30 -0400
Subject: [App_rpt-users] Lots of questions
In-Reply-To: <8D0FE88D-F163-4A3C-85ED-66CB3431BB56@me.com>
References: <CAM09xEbTCCp9LfwqiFz2xU=Y=Sg+iZFAhukfMaLyVpz5AuKhHQ@mail.gmail.com>
	<CAM09xEaBYxEgUR96N6JMN8-MGs-zw-vcmimdP-ki=xgdRxw=XA@mail.gmail.com>
	<132AFE93-8CC9-4B20-8A92-CE0093736A58@me.com>
	<CAM09xEZtmExqUdCzsoYwXvQsH6GT4hZ=_uBymvn0RCR321drAA@mail.gmail.com>
	<8D0FE88D-F163-4A3C-85ED-66CB3431BB56@me.com>
Message-ID: <CAM09xEYFTJBNOMDKQ+O=V-T7Q=4ZVPYZtGfHrTH4fz1RNxy88g@mail.gmail.com>

OK thanks, from more research it looks like people are using other types of
software with raspberry pi such as "thelinkbox".

Apparare Scientior
Paratus Communicare
-N1XBM
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From bdboyle at bdboyle.com  Thu Sep 12 23:05:09 2013
From: bdboyle at bdboyle.com (Bryan D. Boyle)
Date: Thu, 12 Sep 2013 19:05:09 -0400
Subject: [App_rpt-users] Lots of questions
In-Reply-To: <8D0FE88D-F163-4A3C-85ED-66CB3431BB56@me.com>
References: <CAM09xEbTCCp9LfwqiFz2xU=Y=Sg+iZFAhukfMaLyVpz5AuKhHQ@mail.gmail.com>
	<CAM09xEaBYxEgUR96N6JMN8-MGs-zw-vcmimdP-ki=xgdRxw=XA@mail.gmail.com>
	<132AFE93-8CC9-4B20-8A92-CE0093736A58@me.com>
	<CAM09xEZtmExqUdCzsoYwXvQsH6GT4hZ=_uBymvn0RCR321drAA@mail.gmail.com>
	<8D0FE88D-F163-4A3C-85ED-66CB3431BB56@me.com>
Message-ID: <523248A5.3050306@bdboyle.com>

On 9/12/2013 5:44 PM, Tim Sawyer wrote:
> In a word, no.
> --

Darn..didn't even finish the bag of Redenbacker's...



From tim.sawyer at me.com  Fri Sep 13 00:26:40 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Thu, 12 Sep 2013 17:26:40 -0700
Subject: [App_rpt-users] Lots of questions
In-Reply-To: <523248A5.3050306@bdboyle.com>
References: <CAM09xEbTCCp9LfwqiFz2xU=Y=Sg+iZFAhukfMaLyVpz5AuKhHQ@mail.gmail.com>
	<CAM09xEaBYxEgUR96N6JMN8-MGs-zw-vcmimdP-ki=xgdRxw=XA@mail.gmail.com>
	<132AFE93-8CC9-4B20-8A92-CE0093736A58@me.com>
	<CAM09xEZtmExqUdCzsoYwXvQsH6GT4hZ=_uBymvn0RCR321drAA@mail.gmail.com>
	<8D0FE88D-F163-4A3C-85ED-66CB3431BB56@me.com>
	<523248A5.3050306@bdboyle.com>
Message-ID: <8DFB3653-11F8-4B62-8E52-858A0245272C@me.com>

Better luck next time :-)
--
Tim
:wq

On Sep 12, 2013, at 4:05 PM, Bryan D. Boyle <bdboyle at bdboyle.com> wrote:

> On 9/12/2013 5:44 PM, Tim Sawyer wrote:
>> In a word, no.
>> --
> 
> Darn..didn't even finish the bag of Redenbacker's...
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



From nessenj at jimsoffice.org  Fri Sep 13 21:13:34 2013
From: nessenj at jimsoffice.org (James Nessen)
Date: Fri, 13 Sep 2013 14:13:34 -0700
Subject: [App_rpt-users] DMK Eng USB Devices
Message-ID: <CANBAAEuBg5XDQUmA8kw2tNTfy=2-Mww4N59DixRVT1qhf+OUOQ@mail.gmail.com>

Hi Gang,

I have 2 of the DMK Engineering USB sound cards for app_rpt that I am
looking to sell.  Asking  $55.00/each (includes shipping to the US).
 Please email me OFF LIST if you are interested.

Thanks!

Jim, K6JWN
2165 and others.
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From wj0we at yahoo.com  Sun Sep 15 21:46:23 2013
From: wj0we at yahoo.com (joseph Rowe)
Date: Sun, 15 Sep 2013 14:46:23 -0700 (PDT)
Subject: [App_rpt-users] Interfacing a TKR-820 with a URI
Message-ID: <1379281583.79839.YahooMailNeo@web163805.mail.gq1.yahoo.com>

Hello all,

I, quickly, browsed through the archives, but didn't see this question asked / resolved. I am really hoping that someone has the knowledge and time to help with this.

I am setting up an allstar node and running into things that I just don't know how to do. I need to know how to interface (get the right pins to the right pins) the Kenwood TKR-820 UHF repeater's 15-pin accessory port to the URI's 25-pin port. I really don't know which pins are necessary or which ones connect to which ones. I hope this made sense. Below, I have included the websites for the pin layouts of each of these two machines. Any help would be greatly appreciated.

Kenwood TKR-820 15-Pin Accessory Port
http://www.repeater-builder.com/kenwood/tkr-n20-notes.html


URI (USB Radio Interface)
http://qsl.net/kb9mwr/projects/voip/USB-FOB.pdf


Thank you for your time and 73,

Jowe
WJ0WE
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From N1XBM at amsat.org  Mon Sep 16 14:09:17 2013
From: N1XBM at amsat.org (Robert Newberry)
Date: Mon, 16 Sep 2013 10:09:17 -0400
Subject: [App_rpt-users] CTCSS high pass filter
Message-ID: <CAM09xEZ9vRtVFQi5M6DjyscigDQNiuK7Vza=UFrpAcHu6Ehhsg@mail.gmail.com>

Does the software have the ability to act as a high pass filter? Or should
I plan on a hardware solution on my repeater?

Thank you

Apparare Scientior
Paratus Communicare
-N1XBM
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From nessenj at jimsoffice.org  Mon Sep 16 14:22:32 2013
From: nessenj at jimsoffice.org (James Nessen)
Date: Mon, 16 Sep 2013 07:22:32 -0700
Subject: [App_rpt-users] DMK Eng USB Devices
In-Reply-To: <CANBAAEuBg5XDQUmA8kw2tNTfy=2-Mww4N59DixRVT1qhf+OUOQ@mail.gmail.com>
References: <CANBAAEuBg5XDQUmA8kw2tNTfy=2-Mww4N59DixRVT1qhf+OUOQ@mail.gmail.com>
Message-ID: <CANBAAEsNpcbhSaO5a=tHEETa9d3=gY5=1qXD8qdq0V34OYTsSQ@mail.gmail.com>

The URI's  have been sold.  Thanks!

Jim, K6JWN


On Fri, Sep 13, 2013 at 2:13 PM, James Nessen <nessenj at jimsoffice.org>wrote:

> Hi Gang,
>
> I have 2 of the DMK Engineering USB sound cards for app_rpt that I am
> looking to sell.  Asking  $55.00/each (includes shipping to the US).
>  Please email me OFF LIST if you are interested.
>
> Thanks!
>
> Jim, K6JWN
> 2165 and others.
>
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From tim.sawyer at me.com  Mon Sep 16 15:08:26 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Mon, 16 Sep 2013 08:08:26 -0700
Subject: [App_rpt-users] CTCSS high pass filter
In-Reply-To: <CAM09xEZ9vRtVFQi5M6DjyscigDQNiuK7Vza=UFrpAcHu6Ehhsg@mail.gmail.com>
References: <CAM09xEZ9vRtVFQi5M6DjyscigDQNiuK7Vza=UFrpAcHu6Ehhsg@mail.gmail.com>
Message-ID: <E5FF5F77-E269-4AF4-A24B-8B41A6F20823@me.com>

The software can do it.
--
Tim
:wq

On Sep 16, 2013, at 7:09 AM, Robert Newberry <N1XBM at amsat.org> wrote:

> Does the software have the ability to act as a high pass filter? Or should I plan on a hardware solution on my repeater?
> 
> Thank you
> 
> Apparare Scientior 
> Paratus Communicare
> -N1XBM
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From bill.hurlock at cpcomms.com  Tue Sep 17 11:38:20 2013
From: bill.hurlock at cpcomms.com (Bill Hurlock)
Date: Tue, 17 Sep 2013 11:38:20 +0000
Subject: [App_rpt-users] Multi Node Voter questions
Message-ID: <8F656C03689E074C9EE70EE3343E4B841659508B@CPNYMAIL2.cpcomm.int>

I need to put together a private node network that has 12 separate repeaters with 2 voting sites each. The systems are all co located on one rooftop. The voter paths will be over Ubiquity rocket IP networks.
The questions are,

1.       How many nodes can be run on a single server with each node containing 1 main TX/RX and 2 voting sites?

2.       How do you config the nodes to not be part of the Allstar network?

3.       Is there a recommended PC configuration for a server , i.e. CPU speed, Memory, etc.?

Bill Hurlock
CPCommunications
856-234-1661 Office
856-264-1010<http://www.totalrf.com/> Cell
www.cpcomms.com<http://www.cpcomms.com/>
WA2TQI

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From tim.sawyer at me.com  Tue Sep 17 14:13:11 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Tue, 17 Sep 2013 07:13:11 -0700
Subject: [App_rpt-users] Multi Node Voter questions
In-Reply-To: <8F656C03689E074C9EE70EE3343E4B841659508B@CPNYMAIL2.cpcomm.int>
References: <8F656C03689E074C9EE70EE3343E4B841659508B@CPNYMAIL2.cpcomm.int>
Message-ID: <85404010-00EF-40E4-8615-D3818A364F0E@me.com>

Questions 1 and 3 are related. I have no recommendation but my Intel D525 is supporting 11 RTCM's consisting of 6 repeaters. It also supports Allmon (asterisk manager) polling and a few cron jobs. Top shows 90 to 95% idle. 

Private networks are built with a) node numbers under 2000, b) no iax registration, c) no status reporting and d) the IP address of each of the other nodes is in the nodes stanza.
--
Tim
:wq

On Sep 17, 2013, at 4:38 AM, Bill Hurlock <bill.hurlock at cpcomms.com> wrote:

> I need to put together a private node network that has 12 separate repeaters with 2 voting sites each. The systems are all co located on one rooftop. The voter paths will be over Ubiquity rocket IP networks.
> The questions are,
> 1.       How many nodes can be run on a single server with each node containing 1 main TX/RX and 2 voting sites?
> 2.       How do you config the nodes to not be part of the Allstar network?
> 3.       Is there a recommended PC configuration for a server , i.e. CPU speed, Memory, etc.?
>  
> Bill Hurlock
> CPCommunications
> 856-234-1661 Office
> 856-264-1010 Cell
> www.cpcomms.com
> WA2TQI
>  
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From kj6qfs at gmail.com  Tue Sep 17 15:06:00 2013
From: kj6qfs at gmail.com (Sam Skolfield)
Date: Tue, 17 Sep 2013 08:06:00 -0700
Subject: [App_rpt-users] Multi Node Voter questions
In-Reply-To: <8F656C03689E074C9EE70EE3343E4B841659508B@CPNYMAIL2.cpcomm.int>
References: <8F656C03689E074C9EE70EE3343E4B841659508B@CPNYMAIL2.cpcomm.int>
Message-ID: <CALK+jHHY-uSAd1p3HGv+LPN-rJdEM511CEk-TV8_gKthT2DsuA@mail.gmail.com>

Correct me if I am wrong, but an app_rpt server can only handle one master
timing source. Wouldn't that throw a wrench in the possibility of running
multiple voted nodes on one server?


On Tue, Sep 17, 2013 at 4:38 AM, Bill Hurlock <bill.hurlock at cpcomms.com>wrote:

>  I need to put together a private node network that has 12 separate
> repeaters with 2 voting sites each. The systems are all co located on one
> rooftop. The voter paths will be over Ubiquity rocket IP networks.****
>
> The questions are, ****
>
> **1.       **How many nodes can be run on a single server with each node
> containing 1 main TX/RX and 2 voting sites?****
>
> **2.       **How do you config the nodes to not be part of the Allstar
> network?****
>
> **3.       **Is there a recommended PC configuration for a server , i.e.
> CPU speed, Memory, etc.?****
>
> ** **
>
> *Bill Hurlock*
>
> *CPCommunications*****
>
> *856-234-1661 Office*****
>
> *856-264-1010* <http://www.totalrf.com/>* Cell*****
>
> *www.cpcomms.com* <http://www.cpcomms.com/>****
>
> *WA2TQI*****
>
> ** **
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>


-- 
KJ6QFS
Sam Skolfield
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From tim.sawyer at me.com  Tue Sep 17 16:35:56 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Tue, 17 Sep 2013 09:35:56 -0700
Subject: [App_rpt-users] Multi Node Voter questions
In-Reply-To: <CALK+jHHY-uSAd1p3HGv+LPN-rJdEM511CEk-TV8_gKthT2DsuA@mail.gmail.com>
References: <8F656C03689E074C9EE70EE3343E4B841659508B@CPNYMAIL2.cpcomm.int>
	<CALK+jHHY-uSAd1p3HGv+LPN-rJdEM511CEk-TV8_gKthT2DsuA@mail.gmail.com>
Message-ID: <05228BFC-24A5-4A11-B020-0B72B1E62DC6@me.com>

One master can support one or more nodes, each node with one or more clients.  The master must be on the node with the RTCM that is on the same ethernet segment as the server. Here's a sample config...

[general]

port = 667
password = xxxxxxx

[2521]
Blueridge = aaaa1,transmit
buflen = 500

[2529]
SanJuan = xxxx1,transmit
buflen = 500
plfilter = yes

[2531]
Santiago-UHF = rxxxy1,transmit
Lukens-UHF = rxxxy2,transmit
Sunset-UHF = rxxxy3,transmit
PV-UHF = rxxxy4,transmit
TestUHF = rxxxy5,nodeemp
buflen = 180
thresholds = 255,110=5
linger=15
txctcss = 136.5
txctcsslevel = 55
txtoctype = notone
;streams = 184.154.228.18:1667
;streams = wd6awp.net:1667
plfilter = yes

[2532]
Santiago = yyy1,transmit
Huntington = yyy2,master,transmit
PV = yyy3
Woodfern = yyy4
TestBench = yyy5,transmit
buflen = 200
thresholds = 255,110=5
linger=15
plfilter = yes

--
Tim
:wq

On Sep 17, 2013, at 8:06 AM, Sam Skolfield <kj6qfs at gmail.com> wrote:

> Correct me if I am wrong, but an app_rpt server can only handle one master timing source. Wouldn't that throw a wrench in the possibility of running multiple voted nodes on one server?
> 
> 
> On Tue, Sep 17, 2013 at 4:38 AM, Bill Hurlock <bill.hurlock at cpcomms.com> wrote:
> I need to put together a private node network that has 12 separate repeaters with 2 voting sites each. The systems are all co located on one rooftop. The voter paths will be over Ubiquity rocket IP networks.
> 
> The questions are,
> 
> 1.       How many nodes can be run on a single server with each node containing 1 main TX/RX and 2 voting sites?
> 
> 2.       How do you config the nodes to not be part of the Allstar network?
> 
> 3.       Is there a recommended PC configuration for a server , i.e. CPU speed, Memory, etc.?
> 
>  
> 
> Bill Hurlock
> 
> CPCommunications
> 
> 856-234-1661 Office
> 
> 856-264-1010 Cell
> 
> www.cpcomms.com
> 
> WA2TQI
> 
>  
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> 
> 
> 
> -- 
> KJ6QFS
> Sam Skolfield
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From bill.hurlock at cpcomms.com  Tue Sep 17 16:42:21 2013
From: bill.hurlock at cpcomms.com (Bill Hurlock)
Date: Tue, 17 Sep 2013 16:42:21 +0000
Subject: [App_rpt-users] Multi Node Voter questions
In-Reply-To: <85404010-00EF-40E4-8615-D3818A364F0E@me.com>
References: <8F656C03689E074C9EE70EE3343E4B841659508B@CPNYMAIL2.cpcomm.int>
	<85404010-00EF-40E4-8615-D3818A364F0E@me.com>
Message-ID: <8F656C03689E074C9EE70EE3343E4B841659547F@CPNYMAIL2.cpcomm.int>

Couple questions.

1.       In the Voter.conf file stanza. How do I handle the port= # as it relates to each of the nodes RTCMs. I assume they all stay on 667 and the login is based on the RTCM's naming convention i.e. sys1,sys2..... as to what node it becomes associated with as shown below.

2.       As long as everything is on the same private LAN than I only need ONE Master for all nodes on that server correct?

3.       I'm assuming as long as I'm connected to the same private network, I can still use IAXrpt to connect to each of the nodes as long as I list the correct IP addr in the Host field in IAXrtp and the correct user name and REM out the register- statement.

4.       Your Allmon would run off of a small Apachie server running somewhere on the private system.

I'm sure I'll have a couple more questions.
I just read your second reply and note that you have another Master in the 2532 node. I assume that is on a different LAN segment then the other nodes.

[general]
port = 667
buflen =180
password =xxxx

[1998]
Main Site = sys1,master,transmit
Remote 1 = sys2
Remote 2 = sys3

[1999]
Main Site = sys4,transmit
Remote 1 = sys5
Remote 2 = sys6

Bill Hurlock

From: Tim Sawyer [mailto:tim.sawyer at me.com]
Sent: Tuesday, September 17, 2013 10:13 AM
To: Bill Hurlock
Cc: app_rpt-users at ohnosec.org
Subject: Re: [App_rpt-users] Multi Node Voter questions

Questions 1 and 3 are related. I have no recommendation but my Intel D525 is supporting 11 RTCM's consisting of 6 repeaters. It also supports Allmon (asterisk manager) polling and a few cron jobs. Top shows 90 to 95% idle.

Private networks are built with a) node numbers under 2000, b) no iax registration, c) no status reporting and d) the IP address of each of the other nodes is in the nodes stanza.
--
Tim
:wq

On Sep 17, 2013, at 4:38 AM, Bill Hurlock <bill.hurlock at cpcomms.com<mailto:bill.hurlock at cpcomms.com>> wrote:


I need to put together a private node network that has 12 separate repeaters with 2 voting sites each. The systems are all co located on one rooftop. The voter paths will be over Ubiquity rocket IP networks.
The questions are,
1.       How many nodes can be run on a single server with each node containing 1 main TX/RX and 2 voting sites?
2.       How do you config the nodes to not be part of the Allstar network?
3.       Is there a recommended PC configuration for a server , i.e. CPU speed, Memory, etc.?

Bill Hurlock
CPCommunications
856-234-1661 Office
856-264-1010<http://www.totalrf.com/> Cell
www.cpcomms.com<http://www.cpcomms.com/>
WA2TQI

_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org<mailto:App_rpt-users at ohnosec.org>
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From kb2ear at kb2ear.net  Tue Sep 17 17:30:20 2013
From: kb2ear at kb2ear.net (Scott Weis)
Date: Tue, 17 Sep 2013 13:30:20 -0400
Subject: [App_rpt-users] Multi Node Voter questions
In-Reply-To: <05228BFC-24A5-4A11-B020-0B72B1E62DC6@me.com>
References: <8F656C03689E074C9EE70EE3343E4B841659508B@CPNYMAIL2.cpcomm.int>	<CALK+jHHY-uSAd1p3HGv+LPN-rJdEM511CEk-TV8_gKthT2DsuA@mail.gmail.com>
	<05228BFC-24A5-4A11-B020-0B72B1E62DC6@me.com>
Message-ID: <002701ceb3cb$95b03850$c110a8f0$@net>

And you should be able to share the same GPS for all RTCMs that are
co-located at the same sites.

 

From: app_rpt-users-bounces at ohnosec.org
[mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Tim Sawyer
Sent: Tuesday, September 17, 2013 12:36 PM
To: Sam Skolfield
Cc: Bill Hurlock; app_rpt-users at ohnosec.org
Subject: Re: [App_rpt-users] Multi Node Voter questions

 

One master can support one or more nodes, each node with one or more
clients.  The master must be on the node with the RTCM that is on the same
ethernet segment as the server. Here's a sample config...

 

[general]

 

port = 667

password = xxxxxxx

 

[2521]

Blueridge = aaaa1,transmit

buflen = 500

 

[2529]

SanJuan = xxxx1,transmit

buflen = 500

plfilter = yes

 

[2531]

Santiago-UHF = rxxxy1,transmit

Lukens-UHF = rxxxy2,transmit

Sunset-UHF = rxxxy3,transmit

PV-UHF = rxxxy4,transmit

TestUHF = rxxxy5,nodeemp

buflen = 180

thresholds = 255,110=5

linger=15

txctcss = 136.5

txctcsslevel = 55

txtoctype = notone

;streams = 184.154.228.18:1667

;streams = wd6awp.net:1667

plfilter = yes

 

[2532]

Santiago = yyy1,transmit

Huntington = yyy2,master,transmit

PV = yyy3

Woodfern = yyy4

TestBench = yyy5,transmit

buflen = 200

thresholds = 255,110=5

linger=15

plfilter = yes

 

--
Tim
:wq

 

On Sep 17, 2013, at 8:06 AM, Sam Skolfield <kj6qfs at gmail.com> wrote:





Correct me if I am wrong, but an app_rpt server can only handle one master
timing source. Wouldn't that throw a wrench in the possibility of running
multiple voted nodes on one server?

 

On Tue, Sep 17, 2013 at 4:38 AM, Bill Hurlock <bill.hurlock at cpcomms.com>
wrote:

I need to put together a private node network that has 12 separate repeaters
with 2 voting sites each. The systems are all co located on one rooftop. The
voter paths will be over Ubiquity rocket IP networks.

The questions are, 

1.       How many nodes can be run on a single server with each node
containing 1 main TX/RX and 2 voting sites?

2.       How do you config the nodes to not be part of the Allstar network?

3.       Is there a recommended PC configuration for a server , i.e. CPU
speed, Memory, etc.?

 

Bill Hurlock

CPCommunications

856-234-1661 Office

 <http://www.totalrf.com/> 856-264-1010 Cell

 <http://www.cpcomms.com/> www.cpcomms.com

WA2TQI

 


_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users





 

-- 
KJ6QFS
Sam Skolfield 

_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

 

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From cypresstower at yahoo.com  Tue Sep 17 19:21:48 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Tue, 17 Sep 2013 12:21:48 -0700 (PDT)
Subject: [App_rpt-users] I link commands
Message-ID: <1379445708.23647.YahooMailNeo@web163603.mail.gq1.yahoo.com>

Knowingly, the following is found?in the app_rpt.c?under ilink cmdc:??
* 200 thru 215 - (Send DTMF 0-9,*,#,A-D) (200=0, 201=1, 210=*, etc)
Kindly, could?anyone elaborate as to what is the idea behind these functions? 
'google hasn't been very friendly'
JK
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From bill.hurlock at cpcomms.com  Tue Sep 17 19:39:51 2013
From: bill.hurlock at cpcomms.com (Bill Hurlock)
Date: Tue, 17 Sep 2013 19:39:51 +0000
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
Message-ID: <8F656C03689E074C9EE70EE3343E4B841659577D@CPNYMAIL2.cpcomm.int>

I have 2 nodes running on one server. How do I configure IAXrpt to be directed to the right node on the server. One node is 1989 and the other 1990.

Bill Hurlock
CPCommunications
856-234-1661 Office
856-264-1010<http://www.totalrf.com/> Cell
www.cpcomms.com<http://www.cpcomms.com/>
WA2TQI

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From ars.w5omr at gmail.com  Tue Sep 17 19:55:40 2013
From: ars.w5omr at gmail.com (Geoff Edmonson)
Date: Tue, 17 Sep 2013 14:55:40 -0500
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
Message-ID: <vaeetxv189bel8u7k3ugb3i4.1379447740487@email.android.com>

I believe Tim said that while running private nodes, under 2000 as the node number, you couldn't run iax.conf  (no need to register private nodes)


Bill Hurlock <bill.hurlock at cpcomms.com> wrote:

>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From ars.w5omr at gmail.com  Tue Sep 17 19:58:05 2013
From: ars.w5omr at gmail.com (Geoff Edmonson)
Date: Tue, 17 Sep 2013 14:58:05 -0500
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
Message-ID: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>

In fact...

" Private networks are built with a) node numbers under 2000, b) no iax registration, c) no status reporting and d) the IP address of each of the other nodes is in the nodes stanza."




Bill Hurlock <bill.hurlock at cpcomms.com> wrote:

>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From k0jsc.jeff at gmail.com  Tue Sep 17 19:59:36 2013
From: k0jsc.jeff at gmail.com (Jeff Carrier)
Date: Tue, 17 Sep 2013 13:59:36 -0600
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
In-Reply-To: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>
References: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>
Message-ID: <CAA_aZLjRCo8S4MzFgu-4PpwmFdqao_yRvStSVFQy4nFCNd9npA@mail.gmail.com>

which has nothing to do with IaxRPT


On Tue, Sep 17, 2013 at 1:58 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:

> In fact...
>
> " Private networks are built with a) node numbers under 2000, b) no iax
> registration, c) no status reporting and d) the IP address of each of the
> other nodes is in the nodes stanza."
>
>
>
>
> Bill Hurlock <bill.hurlock at cpcomms.com> wrote:
>
>  I have 2 nodes running on one server. How do I configure IAXrpt to be
> directed to the right node on the server. One node is 1989 and the other
> 1990.****
>
> ** **
>
> *Bill Hurlock*
>
> *CPCommunications*****
>
> *856-234-1661 Office*****
>
> *856-264-1010* <http://www.totalrf.com/>* Cell*****
>
> *www.cpcomms.com* <http://www.cpcomms.com/>****
>
> *WA2TQI*****
>
> ** **
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
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From k0jsc.jeff at gmail.com  Tue Sep 17 20:03:13 2013
From: k0jsc.jeff at gmail.com (Jeff Carrier)
Date: Tue, 17 Sep 2013 14:03:13 -0600
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
In-Reply-To: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>
References: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>
Message-ID: <CAA_aZLgaG2Y-9ohnrrh0_cN=vgEdKhT-Z3LtJ8deRN_VnScPdg@mail.gmail.com>

http://www.xelatec.com/xipar/iaxrpt


On Tue, Sep 17, 2013 at 1:58 PM, Geoff Edmonson <ars.w5omr at gmail.com> wrote:

> In fact...
>
> " Private networks are built with a) node numbers under 2000, b) no iax
> registration, c) no status reporting and d) the IP address of each of the
> other nodes is in the nodes stanza."
>
>
>
>
> Bill Hurlock <bill.hurlock at cpcomms.com> wrote:
>
>  I have 2 nodes running on one server. How do I configure IAXrpt to be
> directed to the right node on the server. One node is 1989 and the other
> 1990.****
>
> ** **
>
> *Bill Hurlock*
>
> *CPCommunications*****
>
> *856-234-1661 Office*****
>
> *856-264-1010* <http://www.totalrf.com/>* Cell*****
>
> *www.cpcomms.com* <http://www.cpcomms.com/>****
>
> *WA2TQI*****
>
> ** **
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
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From wb3awj at comcast.net  Tue Sep 17 20:16:43 2013
From: wb3awj at comcast.net (Robert A. Poff WB3AWJ)
Date: Tue, 17 Sep 2013 20:16:43 +0000 (UTC)
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
In-Reply-To: <CAA_aZLgaG2Y-9ohnrrh0_cN=vgEdKhT-Z3LtJ8deRN_VnScPdg@mail.gmail.com>
References: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>
	<CAA_aZLgaG2Y-9ohnrrh0_cN=vgEdKhT-Z3LtJ8deRN_VnScPdg@mail.gmail.com>
Message-ID: <401068760.4675841.1379449003640.JavaMail.root@comcast.net>

Wouldn't you do something like this (?) : 

In iax.conf : 

[iaxRpt_1] 
secret=foo_bar 
type=friend 
host=dynamic 
disallow=alliax.conf : 
iax.conf : 

[iaxRpt_1] 
secret=foo_baiax.conf : 

[iaxRpt_1] 
secret=foo_bar 
type=friend 
host=dynamic 
disallow=all 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=host 
context=radio-gui_1 
requirecalltoken=no 
jbenable=yes 

[iaxRpt_2] 
secret=bar_foo 
type=friend 
host=dynamic 
disallow=all 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=host 
context=radio-gui_2 
requirecalltoken=no 
jbenable=yes 

iax.conf : 

[iaxRpt_1] 
secret=foo_bar 
type=friend 
host=dynamic 
disallow=all 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=hostiax.conf : 

[iaxRpt_1] 
secret=foo_bar 
type=friend 
host=dynamic 
disallow=all 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=host 
context=radio-gui_1 
requirecalltoken=no 
jbenable=yes 

[iaxRpt_2] 
secret=bar_foo 
type=friend 
host=dynamic 
disallow=all 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=host 
context=radio-gui_2 
requirecalltoken=no 
jbenable=yes 


iax.conf : 

[iaxRpt_1] 
secret=foo_bar 
type=friend 
host=dynamic 
disallow=all 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=host 
context=radio-gui_1 
requirecalltoken=no 
jbenable=yes 

[iaxRpt_2] 
secret=bar_foo 
type=friend 
host=dynamic 
disallow=all 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=host 
context=radio-gui_2 
requirecalltoken=no 
jbenable=yes 





extensions.conf : 

[radio-gui_1] 
exten => 27783,1,rpt(27783|X) 
[radio-gui_2] 
exten => 27826,1,rpt(27826|X) 


extensions.conf : 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=host 
context=radio-gui_1 
requirecalltoken=no 
jbenable=yes 

[iaxRpt_2] 
secret=bar_foo 
type=friend 
host=dynamic 
disallow=all 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=host 
context=radio-gui_2 
requirecalltoken=no 
jbenable=yes 

In extensions.conf : 

[radio-gui_1] 
exten => 27783,1,rpt(27783|X) 
[radio-gui_2] 
exten => 27826,1,rpt(27826|X) 
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From wb3awj at comcast.net  Tue Sep 17 20:25:54 2013
From: wb3awj at comcast.net (Robert A. Poff WB3AWJ)
Date: Tue, 17 Sep 2013 20:25:54 +0000 (UTC)
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
In-Reply-To: <401068760.4675841.1379449003640.JavaMail.root@comcast.net>
References: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>
	<CAA_aZLgaG2Y-9ohnrrh0_cN=vgEdKhT-Z3LtJ8deRN_VnScPdg@mail.gmail.com>
	<401068760.4675841.1379449003640.JavaMail.root@comcast.net>
Message-ID: <176549373.4679975.1379449554311.JavaMail.root@comcast.net>

Or a cleaner way..... 

In iax.conf : 

[iaxRpt](!) 
type=friend 
host=dynamic 
disallow=all 
allow=ulaw 
allow=gsm 
allow=g726aal2 
codecpriority=host 
requirecalltoken=no 
jbenable=yes 

[node_A](iaxRpt) 
secret=foo_bar 
context=radio-gui_1 

[node_B](iaxRpt) 
secret=bar_foo 
context=radio-gui_2 



In extensions.conf : 

[radio-gui_1] 
exten => 27783,1,rpt(27783|X) 
[radio-gui_2] 
exten => 27826,1,rpt(27826|X 
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From bill.hurlock at cpcomms.com  Tue Sep 17 20:35:07 2013
From: bill.hurlock at cpcomms.com (Bill Hurlock)
Date: Tue, 17 Sep 2013 20:35:07 +0000
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
In-Reply-To: <CAA_aZLgaG2Y-9ohnrrh0_cN=vgEdKhT-Z3LtJ8deRN_VnScPdg@mail.gmail.com>
References: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>
	<CAA_aZLgaG2Y-9ohnrrh0_cN=vgEdKhT-Z3LtJ8deRN_VnScPdg@mail.gmail.com>
Message-ID: <8F656C03689E074C9EE70EE3343E4B8416595895@CPNYMAIL2.cpcomm.int>

I believe what he was saying is that you don't want to register IAX with allstar.org and you don't want to enable node status reports to Allstar.org. This has nothing to do with being able to use IAX on a private node.

Bill Hurlock

From: Jeff Carrier [mailto:k0jsc.jeff at gmail.com]
Sent: Tuesday, September 17, 2013 4:03 PM
To: Geoff Edmonson
Cc: Bill Hurlock; app_rpt mailing list
Subject: Re: [App_rpt-users] IAXrtp setup for multiple nodes on one server

http://www.xelatec.com/xipar/iaxrpt

On Tue, Sep 17, 2013 at 1:58 PM, Geoff Edmonson <ars.w5omr at gmail.com<mailto:ars.w5omr at gmail.com>> wrote:
In fact...

" Private networks are built with a) node numbers under 2000, b) no iax registration, c) no status reporting and d) the IP address of each of the other nodes is in the nodes stanza."




Bill Hurlock <bill.hurlock at cpcomms.com<mailto:bill.hurlock at cpcomms.com>> wrote:
I have 2 nodes running on one server. How do I configure IAXrpt to be directed to the right node on the server. One node is 1989 and the other 1990.

Bill Hurlock
CPCommunications
856-234-1661<tel:856-234-1661> Office
856-264-1010<http://www.totalrf.com/> Cell
www.cpcomms.com<http://www.cpcomms.com/>
WA2TQI


_______________________________________________
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From k0jsc.jeff at gmail.com  Tue Sep 17 20:45:24 2013
From: k0jsc.jeff at gmail.com (Jeff Carrier)
Date: Tue, 17 Sep 2013 14:45:24 -0600
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
In-Reply-To: <8F656C03689E074C9EE70EE3343E4B8416595895@CPNYMAIL2.cpcomm.int>
References: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>
	<CAA_aZLgaG2Y-9ohnrrh0_cN=vgEdKhT-Z3LtJ8deRN_VnScPdg@mail.gmail.com>
	<8F656C03689E074C9EE70EE3343E4B8416595895@CPNYMAIL2.cpcomm.int>
Message-ID: <CAA_aZLi7KyGdzysOFFw9mFvg5F0X7wi4jRu6ZM=zjsSjUZH23Q@mail.gmail.com>

You are correct.  Your node(s) will still use IAX they just won't register
with the allstar server on the WAN.  You "should" have at least one public
node on the server because of eula and terms of service for the allstar
system.


On Tue, Sep 17, 2013 at 2:35 PM, Bill Hurlock <bill.hurlock at cpcomms.com>wrote:

>  I believe what he was saying is that you don?t want to register IAX with
> allstar.org and you don?t want to enable node status reports to
> Allstar.org. This has nothing to do with being able to use IAX on a private
> node.****
>
> ** **
>
> *Bill Hurlock*
>
> ** **
>
> *From:* Jeff Carrier [mailto:k0jsc.jeff at gmail.com]
> *Sent:* Tuesday, September 17, 2013 4:03 PM
> *To:* Geoff Edmonson
> *Cc:* Bill Hurlock; app_rpt mailing list
> *Subject:* Re: [App_rpt-users] IAXrtp setup for multiple nodes on one
> server****
>
> ** **
>
> http://www.xelatec.com/xipar/iaxrpt****
>
> ** **
>
> On Tue, Sep 17, 2013 at 1:58 PM, Geoff Edmonson <ars.w5omr at gmail.com>
> wrote:****
>
> In fact...
>
> " Private networks are built with a) node numbers under 2000, b) no iax
> registration, c) no status reporting and d) the IP address of each of the
> other nodes is in the nodes stanza."
>
>
>
>
> Bill Hurlock <bill.hurlock at cpcomms.com> wrote:****
>
> I have 2 nodes running on one server. How do I configure IAXrpt to be
> directed to the right node on the server. One node is 1989 and the other
> 1990.****
>
>  ****
>
> *Bill Hurlock*****
>
> *CPCommunications*****
>
> *856-234-1661 Office*****
>
> *856-264-1010* <http://www.totalrf.com/>* Cell*****
>
> *www.cpcomms.com* <http://www.cpcomms.com/>****
>
> *WA2TQI*****
>
>  ****
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users****
>
> ** **
>
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From bill.hurlock at cpcomms.com  Tue Sep 17 23:17:27 2013
From: bill.hurlock at cpcomms.com (Bill Hurlock)
Date: Tue, 17 Sep 2013 23:17:27 +0000
Subject: [App_rpt-users] Multi Node Voter questions
In-Reply-To: <85404010-00EF-40E4-8615-D3818A364F0E@me.com>
References: <8F656C03689E074C9EE70EE3343E4B841659508B@CPNYMAIL2.cpcomm.int>
	<85404010-00EF-40E4-8615-D3818A364F0E@me.com>
Message-ID: <8F656C03689E074C9EE70EE3343E4B8416595BB2@CPNYMAIL2.cpcomm.int>

Just wanted to say thanks for every ones help and thoughts on getting my multimode system up and running. I have all the conf files figured out and I'm now able to connect to each node with IAXrtp. I don't currently have enough RTCM's to complete my big project but I now understand the basics of getting multiple nodes running on one server. I'll still need a little help down the line but I think the big hurtle is behind me for now.

Bill Hurlock
CPCommunications
856-234-1661 Office
856-264-1010<http://www.totalrf.com/> Cell
www.cpcomms.com<http://www.cpcomms.com/>
WA2TQI

_______________________________________________
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From brett.friermood at gmail.com  Wed Sep 18 00:04:11 2013
From: brett.friermood at gmail.com (Brett Friermood)
Date: Tue, 17 Sep 2013 19:04:11 -0500
Subject: [App_rpt-users] I link commands
In-Reply-To: <1379445708.23647.YahooMailNeo@web163603.mail.gq1.yahoo.com>
References: <1379445708.23647.YahooMailNeo@web163603.mail.gq1.yahoo.com>
Message-ID: <CADS0AtSx+Gf54DV933xgfs-fJCJqfQS4LiGfxtO85DQG8=Ou=A@mail.gmail.com>

I believe those send actual DTMF tones.

ilink,200 sends an actual DTMF "0"
...up thru...
ilink,209 sends an actual DTMF "9"

And then incrementing through *,#,A,B,C

Ending with ilink,215 sending an actual DTMF "D"

Brett KQ9N

On Tue, Sep 17, 2013 at 2:21 PM, Johnny Keeker <cypresstower at yahoo.com> wrote:
> Knowingly, the following is found in the app_rpt.c under ilink cmdc:
> * 200 thru 215 - (Send DTMF 0-9,*,#,A-D) (200=0, 201=1, 210=*, etc)
> Kindly, could anyone elaborate as to what is the idea behind these
> functions?
> 'google hasn't been very friendly'
> JK
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>


From tim.sawyer at me.com  Wed Sep 18 03:17:20 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Tue, 17 Sep 2013 20:17:20 -0700
Subject: [App_rpt-users] IAXrtp setup for multiple nodes on one server
In-Reply-To: <CAA_aZLi7KyGdzysOFFw9mFvg5F0X7wi4jRu6ZM=zjsSjUZH23Q@mail.gmail.com>
References: <scruqimtvtfk1bjot3n58v3v.1379447885052@email.android.com>
	<CAA_aZLgaG2Y-9ohnrrh0_cN=vgEdKhT-Z3LtJ8deRN_VnScPdg@mail.gmail.com>
	<8F656C03689E074C9EE70EE3343E4B8416595895@CPNYMAIL2.cpcomm.int>
	<CAA_aZLi7KyGdzysOFFw9mFvg5F0X7wi4jRu6ZM=zjsSjUZH23Q@mail.gmail.com>
Message-ID: <AE1D3F94-0B8D-4D38-9FC1-392F7FEF8918@me.com>

Could you show us where that eula is please.  
--
Tim
:wq

On Sep 17, 2013, at 1:45 PM, Jeff Carrier <k0jsc.jeff at gmail.com> wrote:

> You "should" have at least one public node on the server because of eula and terms of service for the allstar system.

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From kc7mrq at bresnan.net  Wed Sep 18 20:00:53 2013
From: kc7mrq at bresnan.net (Corey Badgley)
Date: Wed, 18 Sep 2013 14:00:53 -0600
Subject: [App_rpt-users] URI PTT issue
Message-ID: <523A0675.3080605@bresnan.net>

I have an URI on port three of an SCOM 7330 controller. Everything works 
perfectly unless power goes out and the node server reboots. The URI 
keys the controller while the computer boots and stays keyed until one 
of the following occurs:

1. Audio from another node is received.
2. Telemetry is broadcasted. ( I usually have telemetry turned off)
3. Send a command to un-key the PTT.
4. Enter echo mode within radio-tune-menu and my audio is rebroadcasted.

Is there a way to toggle the URI PTT upon a reboot or start-up to ensure 
its logic is in the correct state?


Have a good day,

Corey


From keith at goobie.org  Wed Sep 18 20:05:05 2013
From: keith at goobie.org (Keith Goobie)
Date: Wed, 18 Sep 2013 16:05:05 -0400
Subject: [App_rpt-users] URI PTT issue
In-Reply-To: <523A0675.3080605@bresnan.net>
Message-ID: <CE5F7FB1.2362A%keith@goobie.org>

Why not program the SCOM to use an inverted PTT signal and program the
usbradio.conf to use an inverted signal for PTT.  That should deal with the
re-boot issue.

Keith


On 9/18/13 4:00 PM, "Corey Badgley" <kc7mrq at bresnan.net> wrote:

> I have an URI on port three of an SCOM 7330 controller. Everything works
> perfectly unless power goes out and the node server reboots. The URI
> keys the controller while the computer boots and stays keyed until one
> of the following occurs:
> 
> 1. Audio from another node is received.
> 2. Telemetry is broadcasted. ( I usually have telemetry turned off)
> 3. Send a command to un-key the PTT.
> 4. Enter echo mode within radio-tune-menu and my audio is rebroadcasted.
> 
> Is there a way to toggle the URI PTT upon a reboot or start-up to ensure
> its logic is in the correct state?
> 
> 
> Have a good day,
> 
> Corey
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

-- 
keith at goobie.org
Keith Goobie
Richmond Hill, ON, CANADA




From tim.sawyer at me.com  Thu Sep 19 00:28:26 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Wed, 18 Sep 2013 17:28:26 -0700
Subject: [App_rpt-users] URI PTT issue
In-Reply-To: <CE5F7FB1.2362A%keith@goobie.org>
References: <CE5F7FB1.2362A%keith@goobie.org>
Message-ID: <E9F7A551-3703-4706-B323-3564E52406AE@me.com>

Also be sure the PTT-2-COR line is pulled up.
--
Tim
:wq

On Sep 18, 2013, at 1:05 PM, Keith Goobie <keith at goobie.org> wrote:

> Why not program the SCOM to use an inverted PTT signal and program the
> usbradio.conf to use an inverted signal for PTT.  That should deal with the
> re-boot issue.
> 
> Keith
> 
> 
> On 9/18/13 4:00 PM, "Corey Badgley" <kc7mrq at bresnan.net> wrote:
> 
>> I have an URI on port three of an SCOM 7330 controller. Everything works
>> perfectly unless power goes out and the node server reboots. The URI
>> keys the controller while the computer boots and stays keyed until one
>> of the following occurs:
>> 
>> 1. Audio from another node is received.
>> 2. Telemetry is broadcasted. ( I usually have telemetry turned off)
>> 3. Send a command to un-key the PTT.
>> 4. Enter echo mode within radio-tune-menu and my audio is rebroadcasted.
>> 
>> Is there a way to toggle the URI PTT upon a reboot or start-up to ensure
>> its logic is in the correct state?
>> 
>> 
>> Have a good day,
>> 
>> Corey
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> -- 
> keith at goobie.org
> Keith Goobie
> Richmond Hill, ON, CANADA
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



From kuggie at kuggie.com  Thu Sep 19 01:49:38 2013
From: kuggie at kuggie.com (Kevin Custer)
Date: Wed, 18 Sep 2013 21:49:38 -0400
Subject: [App_rpt-users] URI PTT issue
In-Reply-To: <CE5F7FB1.2362A%keith@goobie.org>
References: <CE5F7FB1.2362A%keith@goobie.org>
Message-ID: <523A5832.7080809@kuggie.com>

Or, consider a different radio interface - one that doesn't let the PTT 
happen unless flashy-flashy is going on...

http://www.repeater-builder.com/products/usb-rim.html

Disclaimer:
I have no financial interest in this product - however, I did help 
design it.

Kevin Custer - WJ8G



On 9/18/2013 4:05 PM, Keith Goobie wrote:
> Why not program the SCOM to use an inverted PTT signal and program the
> usbradio.conf to use an inverted signal for PTT.  That should deal with the
> re-boot issue.
>
> Keith
>
>
> On 9/18/13 4:00 PM, "Corey Badgley" <kc7mrq at bresnan.net> wrote:
>
>> I have an URI on port three of an SCOM 7330 controller. Everything works
>> perfectly unless power goes out and the node server reboots. The URI
>> keys the controller while the computer boots and stays keyed until one
>> of the following occurs:
>>
>> 1. Audio from another node is received.
>> 2. Telemetry is broadcasted. ( I usually have telemetry turned off)
>> 3. Send a command to un-key the PTT.
>> 4. Enter echo mode within radio-tune-menu and my audio is rebroadcasted.
>>
>> Is there a way to toggle the URI PTT upon a reboot or start-up to ensure
>> its logic is in the correct state?
>>
>>
>> Have a good day,
>>
>> Corey



From keith at goobie.org  Thu Sep 19 12:47:04 2013
From: keith at goobie.org (Keith Goobie)
Date: Thu, 19 Sep 2013 08:47:04 -0400
Subject: [App_rpt-users] RELAXED RADIO parms
Message-ID: <CE606A88.23679%keith@goobie.org>

Good morning All.

I am looking for the message that Jim WB6NIL put out there that had to do
with modifying the dsp.c code so that it did not incorporate the
RELAXED_RADIO parameters.

One parameter is the length of time that a DTMF signal must be present
before it is determined not to be a false signal.
[root at Server6 main]# diff ?dsp.c.default ?dsp.c
>> 165c165,166
>> < #define DTMF_TO_TOTAL_ENERGY ? ?((digitmode & DSP_DIGITMODE_RELAXDTMF) ?
>> 26.0 : 42.0)
>> ---
> #define DTMF_TO_TOTAL_ENERGY ? ?((digitmode & DSP_DIGITMODE_RELAXDTMF) ? 38.0
: 42.0) ?/* CAH changed 26 to 38 */

This change when implemented, did eliminate a noticeable number of audio
pops associated with DTMF false-ing.

The message from Jim spoke to bypassing the RELAXED parameters in total.

I am going through  the archives and I have not found it yet.

Thanks in advance.

Keith

-- 
keith at goobie.org
Keith Goobie
Richmond Hill, ON, CANADA

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From dshaw at ke6upi.com  Thu Sep 19 14:13:42 2013
From: dshaw at ke6upi.com (David KE6UPI)
Date: Thu, 19 Sep 2013 07:13:42 -0700
Subject: [App_rpt-users] RELAXED RADIO parms
In-Reply-To: <CE606A88.23679%keith@goobie.org>
References: <CE606A88.23679%keith@goobie.org>
Message-ID: <CAH4Rvb+MP4xo-nE7LSA3j_axuOzkdXJnD6i3ONYXNy4z5mgmbg@mail.gmail.com>

look for:

Post Date 6/13/12
Subject RTCM falsing DTMF

Please read it all !!!!

David

On Thu, Sep 19, 2013 at 5:47 AM, Keith Goobie <keith at goobie.org> wrote:

>  Good morning All.
>
> I am looking for the message that Jim WB6NIL put out there that had to do
> with modifying the dsp.c code so that it did not incorporate the
> RELAXED_RADIO parameters.
>
> One parameter is the length of time that a DTMF signal must be present
> before it is determined not to be a false signal.
> [root at Server6 main]# diff  dsp.c.default  dsp.c
>
> 165c165,166
> < #define DTMF_TO_TOTAL_ENERGY    ((digitmode & DSP_DIGITMODE_RELAXDTMF) ?
> 26.0 : 42.0)
> ---
>
> > #define DTMF_TO_TOTAL_ENERGY    ((digitmode & DSP_DIGITMODE_RELAXDTMF) ?
> 38.0 : 42.0)  /* CAH changed 26 to 38 */
>
> This change when implemented, did eliminate a noticeable number of audio
> pops associated with DTMF false-ing.
>
> The message from Jim spoke to bypassing the RELAXED parameters in total.
>
> I am going through  the archives and I have not found it yet.
>
> Thanks in advance.
>
> Keith
>
> --
> keith at goobie.org
> Keith Goobie
> Richmond Hill, ON, CANADA
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
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From keith at goobie.org  Thu Sep 19 14:24:10 2013
From: keith at goobie.org (Keith Goobie)
Date: Thu, 19 Sep 2013 10:24:10 -0400
Subject: [App_rpt-users] RELAXED RADIO parms
In-Reply-To: <CAH4Rvb+MP4xo-nE7LSA3j_axuOzkdXJnD6i3ONYXNy4z5mgmbg@mail.gmail.com>
Message-ID: <CE60814A.23687%keith@goobie.org>

Thanks, will do.

Keith


On 9/19/13 10:13 AM, "David KE6UPI" <dshaw at ke6upi.com> wrote:

> look for:
> 
> Post Date 6/13/12
> Subject?RTCM falsing DTMF
> 
> Please read it all !!!!
> 
> David
> 
> On Thu, Sep 19, 2013 at 5:47 AM, Keith Goobie <keith at goobie.org> wrote:
>> Good morning All.
>> 
>> I am looking for the message that Jim WB6NIL put out there that had to do
>> with modifying the dsp.c code so that it did not incorporate the
>> RELAXED_RADIO parameters.
>> 
>> One parameter is the length of time that a DTMF signal must be present before
>> it is determined not to be a false signal. ?
>> [root at Server6 main]# diff ?dsp.c.default ?dsp.c
>>>> 165c165,166
>>>> < #define DTMF_TO_TOTAL_ENERGY ? ?((digitmode & DSP_DIGITMODE_RELAXDTMF) ?
>>>> 26.0 : 42.0)
>>>> ---
>>> > #define DTMF_TO_TOTAL_ENERGY ? ?((digitmode & DSP_DIGITMODE_RELAXDTMF) ?
>>> 38.0 : 42.0) ?/* CAH changed 26 to 38 */
>> 
>> This change when implemented, did eliminate a noticeable number of audio pops
>> associated with DTMF false-ing. ?
>> 
>> The message from Jim spoke to bypassing the RELAXED parameters in total.
>> 
>> I am going through ?the archives and I have not found it yet.
>> 
>> Thanks in advance.
>> 
>> Keith

-- 
keith at goobie.org
Keith Goobie
Richmond Hill, ON, CANADA

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From keith at handscombe.co.uk  Thu Sep 19 16:19:21 2013
From: keith at handscombe.co.uk (keith)
Date: Thu, 19 Sep 2013 17:19:21 +0100
Subject: [App_rpt-users] Node 2498 is no longer in the database
Message-ID: <000001ceb553$ffcea990$ff6bfcb0$@co.uk>

Good evening all,

 

I have noticed that on the (https://allstarlink.org/nodelist.php) site I
show up as node 2498 in green but when I click onto the MB7IIP text I get
the Node 2498 is no longer in the database. I have made no changes on my LAN
and the IP of the node has been a static address since it was built. The
ports on router are configured and for a short time I even set the router to
DMZ the nodes address so no router interactio.

My node works fine. If I reboot it does take about an hour to authenticate
but once authenticated all works great.

 

Any ideas please

 

   Kind Regards

 

   Keith Handscombe     

 

154e886

 

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From keith at goobie.org  Thu Sep 19 16:41:02 2013
From: keith at goobie.org (Keith Goobie)
Date: Thu, 19 Sep 2013 12:41:02 -0400
Subject: [App_rpt-users] Node 2498 is no longer in the database
In-Reply-To: <000001ceb553$ffcea990$ff6bfcb0$@co.uk>
Message-ID: <CE60A15E.236B2%keith@goobie.org>

Keith

As a diagnostic try the following:

validate that in the rpt.conf file that you are updating to the allstar
server
        statpost_url=http://stats.allstarlink.org/uhandler.php ; Status
updates
        
statpost_program=/usr/bin/wget,-q,--timeout=15,--tries=1,--output-document=/
dev/null

Being granular, you can see if you are updating to the stats server, by
typing at the command prompt:
        tcpdump port 80

Every five minutes or so, you will see a blast of data go to the
stats.allstarlink.org server.  If you do not see this activity, your server
either has a broken internet connection, or a firewall rule issue, although
typically as this is outbound it should not be an issue.  The info presented
may be cryptic, but it will validate that the node is doing updates.

There may be a dns issue in play, and you can use nslookup to validate that.
The latest Centos updates did do some hanky panky with the DNS config files
? check for a forwarders artifact that you may not have had before.  Along
the dns issue, check to make sure that ipv6 is not in play.

While you are doing a snoop using tcpdump, you can also validate that you
are in a nodes list published by the allstarlink server and it is downloaded
about every ten minutes on average.  In that file, you should see the info
that the server has on your node.  That file is rpt_extnodes and is located
in the var/lib/asterisk folder.

Can other nodes connect with you?  I suspect that the response is no, but
just checking.

Keith
VA3YC




On 9/19/13 12:19 PM, "keith" <keith at handscombe.co.uk> wrote:

> Good evening all,
>  
> I have noticed that on the (https://allstarlink.org/nodelist.php) site I show
> up as node 2498 in green but when I click onto the MB7IIP text I get the Node
> 2498 is no longer in the database. I have made no changes on my LAN and the IP
> of the node has been a static address since it was built. The ports on router
> are configured and for a short time I even set the router to DMZ the nodes
> address so no router interactio.
> My node works fine. If I reboot it does take about an hour to authenticate but
> once authenticated all works great.
>  
> Any ideas please
>  
>    Kind Regards
>  
>    Keith Handscombe
>  
>  
>  
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

-- 
keith at goobie.org
Keith Goobie
Richmond Hill, ON, CANADA

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From dshaw at ke6upi.com  Thu Sep 19 18:37:56 2013
From: dshaw at ke6upi.com (David KE6UPI)
Date: Thu, 19 Sep 2013 11:37:56 -0700
Subject: [App_rpt-users] Fwd: Jim is famous in Mexico.
In-Reply-To: <CAH4RvbLtt-4598z-6vvJhmE4wt9anJvhww+Tf7NOFqEST6OFZA@mail.gmail.com>
References: <CAH4RvbLtt-4598z-6vvJhmE4wt9anJvhww+Tf7NOFqEST6OFZA@mail.gmail.com>
Message-ID: <CAH4RvbJ0CGBT48H+pD4dwbtQ0NraOd3KX_YZpJW574iveUfvEw@mail.gmail.com>

I found that Jim has a following in Mexico. He is famous there and they
that put a likeness of him on the 10 Nuevos Peso.

Nice going Jim. Keep up the great work..


David
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From kb4fxc at inttek.net  Thu Sep 19 18:49:28 2013
From: kb4fxc at inttek.net (David McGough)
Date: Thu, 19 Sep 2013 14:49:28 -0400 (EDT)
Subject: [App_rpt-users]  Zaptel on Debian Wheezy???
In-Reply-To: <CAH4RvbJ0CGBT48H+pD4dwbtQ0NraOd3KX_YZpJW574iveUfvEw@mail.gmail.com>
Message-ID: <Pine.LNX.4.44.1309191439090.20536-100000@goliath.inttek.net>


Hi Everyone,

Has anyone ported the Zaptel driver library to recent Linux kernels, 
specifically, kernel rev 3.2.x? ....I'm working on Debian Wheezy.

Thanks!

73, David KB4FXC





From telesistant at hotmail.com  Thu Sep 19 18:52:10 2013
From: telesistant at hotmail.com (Jim Duuuude)
Date: Thu, 19 Sep 2013 11:52:10 -0700
Subject: [App_rpt-users] Zaptel on Debian Wheezy???
In-Reply-To: <Pine.LNX.4.44.1309191439090.20536-100000@goliath.inttek.net>
References: <CAH4RvbJ0CGBT48H+pD4dwbtQ0NraOd3KX_YZpJW574iveUfvEw@mail.gmail.com>,
	<Pine.LNX.4.44.1309191439090.20536-100000@goliath.inttek.net>
Message-ID: <BLU180-W305205237319016E0C882BB1210@phx.gbl>

Its not just a 'port'. 3.X COMPLETELY changed a lot of things, in particular
the ioctl() interface. POOP!

Try using DAHDI with the patches that make it 'work' (see list archives for this info).
Its also somewhere on the SVN, I think.

Jim

> Date: Thu, 19 Sep 2013 14:49:28 -0400
> From: kb4fxc at inttek.net
> To: app_rpt-users at ohnosec.org
> Subject: [App_rpt-users]  Zaptel on Debian Wheezy???
> 
> 
> Hi Everyone,
> 
> Has anyone ported the Zaptel driver library to recent Linux kernels, 
> specifically, kernel rev 3.2.x? ....I'm working on Debian Wheezy.
> 
> Thanks!
> 
> 73, David KB4FXC
> 
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
 		 	   		  
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From kb4fxc at inttek.net  Thu Sep 19 19:38:45 2013
From: kb4fxc at inttek.net (David McGough)
Date: Thu, 19 Sep 2013 15:38:45 -0400 (EDT)
Subject: [App_rpt-users] Zaptel on Debian Wheezy???
In-Reply-To: <BLU180-W305205237319016E0C882BB1210@phx.gbl>
Message-ID: <Pine.LNX.4.44.1309191524540.22635-100000@goliath.inttek.net>


Hi Jim,

I realize that a lot has changed with Linux 3.X.  But, the mapping from
the old ioctl() kernel calls to the new compat_ioctl() or unlocked_ioctl()  
calls is pretty straight forward for many ioclt operations......I haven't
carefully looked zaptel over as of yet. Since I only really need ztdummy
and zttranscode, I was hoping a port would be simple--and, already tackled
by someone else!

I have concerns about trying to use the dahdi code in a production 
environment at this point, since it hasn't been as heavily tested (or am I 
wrong about this?).

Thanks for your help-

73, David KB4FXC



On Thu, 19 Sep 2013, Jim Duuuude wrote:

> Its not just a 'port'. 3.X COMPLETELY changed a lot of things, in particular
> the ioctl() interface. POOP!
> 
> Try using DAHDI with the patches that make it 'work' (see list archives for this info).
> Its also somewhere on the SVN, I think.
> 
> Jim
> 
> > Date: Thu, 19 Sep 2013 14:49:28 -0400
> > From: kb4fxc at inttek.net
> > To: app_rpt-users at ohnosec.org
> > Subject: [App_rpt-users]  Zaptel on Debian Wheezy???
> > 
> > 
> > Hi Everyone,
> > 
> > Has anyone ported the Zaptel driver library to recent Linux kernels, 
> > specifically, kernel rev 3.2.x? ....I'm working on Debian Wheezy.
> > 
> > Thanks!
> > 
> > 73, David KB4FXC
> > 
> > 
> > 
> > _______________________________________________
> > App_rpt-users mailing list
> > App_rpt-users at ohnosec.org
> > http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>  		 	   		  



From kb4fxc at inttek.net  Thu Sep 19 20:13:29 2013
From: kb4fxc at inttek.net (David McGough)
Date: Thu, 19 Sep 2013 16:13:29 -0400 (EDT)
Subject: [App_rpt-users] Zaptel on Debian Wheezy???
In-Reply-To: <Pine.LNX.4.44.1309191524540.22635-100000@goliath.inttek.net>
Message-ID: <Pine.LNX.4.44.1309191609010.23528-100000@goliath.inttek.net>


Correction: I just need zaptel.ko and ztdummy.ko, not zttranscode.ko.

On Thu, 19 Sep 2013, David McGough wrote:

> 
> Hi Jim,
> 
> I realize that a lot has changed with Linux 3.X.  But, the mapping from
> the old ioctl() kernel calls to the new compat_ioctl() or unlocked_ioctl()  
> calls is pretty straight forward for many ioclt operations......I haven't
> carefully looked zaptel over as of yet. Since I only really need ztdummy
> and zttranscode, I was hoping a port would be simple--and, already tackled
> by someone else!
> 
> I have concerns about trying to use the dahdi code in a production 
> environment at this point, since it hasn't been as heavily tested (or am I 
> wrong about this?).
> 
> Thanks for your help-
> 
> 73, David KB4FXC
> 
> 
> 
> On Thu, 19 Sep 2013, Jim Duuuude wrote:
> 
> > Its not just a 'port'. 3.X COMPLETELY changed a lot of things, in particular
> > the ioctl() interface. POOP!
> > 
> > Try using DAHDI with the patches that make it 'work' (see list archives for this info).
> > Its also somewhere on the SVN, I think.
> > 
> > Jim
> > 
> > > Date: Thu, 19 Sep 2013 14:49:28 -0400
> > > From: kb4fxc at inttek.net
> > > To: app_rpt-users at ohnosec.org
> > > Subject: [App_rpt-users]  Zaptel on Debian Wheezy???
> > > 
> > > 
> > > Hi Everyone,
> > > 
> > > Has anyone ported the Zaptel driver library to recent Linux kernels, 
> > > specifically, kernel rev 3.2.x? ....I'm working on Debian Wheezy.
> > > 
> > > Thanks!
> > > 
> > > 73, David KB4FXC
> > > 
> > > 
> > > 
> > > _______________________________________________
> > > App_rpt-users mailing list
> > > App_rpt-users at ohnosec.org
> > > http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> >  		 	   		  
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 



From n8ohu at yahoo.com  Fri Sep 20 00:31:37 2013
From: n8ohu at yahoo.com (Matthew)
Date: Thu, 19 Sep 2013 20:31:37 -0400
Subject: [App_rpt-users] Node 2498 is no longer in the database
Message-ID: <b2leponxj4fpfq49ufxgrveb.1379637097066@email.android.com>

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From keith at goobie.org  Fri Sep 20 00:58:09 2013
From: keith at goobie.org (Keith Goobie)
Date: Thu, 19 Sep 2013 20:58:09 -0400
Subject: [App_rpt-users] Node 2498 is no longer in the database
In-Reply-To: <b2leponxj4fpfq49ufxgrveb.1379637097066@email.android.com>
Message-ID: <CE6115E1.236E6%keith@goobie.org>

Matthew

As I pointed out to Keith in the UK, there are a number of networking things
that you should look at.

* validate that updates are going from your node to the allstarlink site
* check that your node is listed in rpt_extnodes
* verify that your dns info is correct and the queries are answered quickly.
* check that the Internet is running correctly.

Keith


On 9/19/13 8:31 PM, "Matthew" <n8ohu at yahoo.com> wrote:

> Keith,
> 
> I'm not sure of the cause, but I've had it happen with my own nodes from time
> to time; most recently I noticed it happen wih node 29822.
> 
> Matthew Pitts
> N8OHU 
> 
> Sent from my Verizon Wireless 4G LTE smartphone
> 
> -------- Original Message --------
> Subject:[App_rpt-users] Node 2498 is no longer in the database
> From :keith 
> Date :Thu, 19-Sep-2013 12:19
> To :app_rpt-users at ohnosec.org
> CC :
> 
> Good evening all,
>  
> I have noticed that on the (https://allstarlink.org/nodelist.php) site I show
> up as node 2498 in green but when I click onto the MB7IIP text I get the Node
> 2498 is no longer in the database. I have made no changes on my LAN and the IP
> of the node has been a static address since it was built. The ports on router
> are configured and for a short time I even set the router to DMZ the nodes
> address so no router interactio.
> My node works fine. If I reboot it does take about an hour to authenticate but
> once authenticated all works great.
>  
> Any ideas please
>  
>    Kind Regards
>  
>    Keith Handscombe
>  
>  
>  
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

-- 
keith at goobie.org
Keith Goobie
Richmond Hill, ON, CANADA

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From specialq.que at ntlworld.com  Fri Sep 20 18:12:32 2013
From: specialq.que at ntlworld.com (DougH)
Date: Fri, 20 Sep 2013 19:12:32 +0100
Subject: [App_rpt-users] Telephone Portal?
Message-ID: <6778B992C697456AABEA0DC16BEC6D0B@DougLaptop2>

Hello,

Trying to use the UK portal, enter node number and #, system says invalid 
selection, asks for number again then after entering once more, says there 
is a problem and discconnects.

What am I not doing correctly?

Doug - GB3KE 2253 



From markjohnston73 at gmail.com  Fri Sep 20 22:05:57 2013
From: markjohnston73 at gmail.com (Mark Johnston)
Date: Fri, 20 Sep 2013 15:05:57 -0700
Subject: [App_rpt-users] Remote base Memories
Message-ID: <CAARNdJHjajsV_V33zuTF-wA-d9n9=BBAwJR6xejau0y+=+0UgQ@mail.gmail.com>

Hello everyone,

I have finally got the ic-706 remote base partially working
I can scan freq, move up in freq, and read status, Now that the commands
also the init=freq..... works when I connect so it can set the frequency.
just fine, yet if I do *1444*000*0... etc does not work
If I do *0001 for memory 001 etc, I get 'memory not found'

which the memory is in the rpt.conf

[memory]
  ;Syntax : CC=RRR.RRR,PPP.P,AAAAA
  ;          C=Channel Number
  ;          R=RX Frequency
  ;          P=PL Frequency
  ;          A=Attributes
  ;
  ;Attributes : a=AM   b=LSB  f=FM  u=USB
  ;             l=Low Power   m=Medium Power   h=High Power
  ;             -=Minus TX Offset    s=Simplex     +=Plus TX Offset
  ;             t=TX PL on    r=RX PL on
  ;
004=147.700,123.0,fshrt
005=147.100,123.0,fshtt
006=003.860,100.0,bshrt
init=003.980,100.0,bshrt ;Set to this on startup


This is just above the macro section below the functions-remote...

At the end of the file I tried[memory]

[memory]
001=147.700,123.0,fshrt
002=147.100,123.0,fshtt
003=003.860,100.0,bshrt


None of these work, always says 'memory not found'

Any ideas? I know I am getting close to making this work!

0=remote,1                             ; Retrieve Memory
1=remote,2

Mark KC7DMF
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From wb3awj at comcast.net  Sat Sep 21 02:21:05 2013
From: wb3awj at comcast.net (Robert A. Poff)
Date: Fri, 20 Sep 2013 22:21:05 -0400
Subject: [App_rpt-users] Remote base Memories
In-Reply-To: <CAARNdJHjajsV_V33zuTF-wA-d9n9=BBAwJR6xejau0y+=+0UgQ@mail.gmail.com>
References: <CAARNdJHjajsV_V33zuTF-wA-d9n9=BBAwJR6xejau0y+=+0UgQ@mail.gmail.com>
Message-ID: <5e28de40-69c0-40b7-9e5b-c0c45a2ec10a@email.android.com>

Try numbering your memories from 00 to 99 instead of three digits. 



Mark Johnston <markjohnston73 at gmail.com> wrote:
>Hello everyone,
>
>I have finally got the ic-706 remote base partially working
>I can scan freq, move up in freq, and read status, Now that the
>commands
>also the init=freq..... works when I connect so it can set the
>frequency.
>just fine, yet if I do *1444*000*0... etc does not work
>If I do *0001 for memory 001 etc, I get 'memory not found'
>
>which the memory is in the rpt.conf
>
>[memory]
>  ;Syntax : CC=RRR.RRR,PPP.P,AAAAA
>  ;          C=Channel Number
>  ;          R=RX Frequency
>  ;          P=PL Frequency
>  ;          A=Attributes
>  ;
>  ;Attributes : a=AM   b=LSB  f=FM  u=USB
>  ;             l=Low Power   m=Medium Power   h=High Power
>  ;             -=Minus TX Offset    s=Simplex     +=Plus TX Offset
>  ;             t=TX PL on    r=RX PL on
>  ;
>004=147.700,123.0,fshrt
>005=147.100,123.0,fshtt
>006=003.860,100.0,bshrt
>init=003.980,100.0,bshrt ;Set to this on startup
>
>
>This is just above the macro section below the functions-remote...
>
>At the end of the file I tried[memory]
>
>[memory]
>001=147.700,123.0,fshrt
>002=147.100,123.0,fshtt
>003=003.860,100.0,bshrt
>
>
>None of these work, always says 'memory not found'
>
>Any ideas? I know I am getting close to making this work!
>
>0=remote,1                             ; Retrieve Memory
>1=remote,2
>
>Mark KC7DMF
>
>
>------------------------------------------------------------------------
>
>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

-- 
Sent from my Android phone with K-9 Mail. Please excuse my brevity and typing errors.
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From markjohnston73 at gmail.com  Sat Sep 21 03:13:16 2013
From: markjohnston73 at gmail.com (Mark Johnston)
Date: Fri, 20 Sep 2013 20:13:16 -0700
Subject: [App_rpt-users] Remote base Memories
In-Reply-To: <5e28de40-69c0-40b7-9e5b-c0c45a2ec10a@email.android.com>
References: <CAARNdJHjajsV_V33zuTF-wA-d9n9=BBAwJR6xejau0y+=+0UgQ@mail.gmail.com>
	<5e28de40-69c0-40b7-9e5b-c0c45a2ec10a@email.android.com>
Message-ID: <CAARNdJHhwazFDg-imzWK19-=7Sw9GmU1t-+xOtzv1oGXCMyzJw@mail.gmail.com>

I tried that, does not work, says cannot find memory still
however when I did *006 it says invalid frequency, and switches to VFO A
from B or memory if I manually put it there...

06 I deleted as a memory channel, and find no ref to in the rest of the
file... as a command
.... very interesting for sure!

anyone have an idea? I know i am so close to making this work.

"Got Root?"

How many software engineers does it take to change a light bulb?
*None. It's a hardware problem.*


Unix is user friendly. It's just very particular about who it's friends are.
WINDOWS: Will Install Needless Data On Whole System
MICROSOFT: Most Intelligent Customers Realize Our Software Only Fools
Teenagers.


A ntennas
P oorly
P laced
L acks
E ngineering

The best way to accelerate a computer running Windows is at 9.81 m/s?.

*"I get paid to support Windows, I use Linux to get work done."*



On Fri, Sep 20, 2013 at 7:21 PM, Robert A. Poff <wb3awj at comcast.net> wrote:

> Try numbering your memories from 00 to 99 instead of three digits.
>
>
>
> Mark Johnston <markjohnston73 at gmail.com> wrote:
>>
>> Hello everyone,
>>
>> I have finally got the ic-706 remote base partially working
>> I can scan freq, move up in freq, and read status, Now that the commands
>> also the init=freq..... works when I connect so it can set the frequency.
>> just fine, yet if I do *1444*000*0... etc does not work
>> If I do *0001 for memory 001 etc, I get 'memory not found'
>>
>> which the memory is in the rpt.conf
>>
>> [memory]
>>   ;Syntax : CC=RRR.RRR,PPP.P,AAAAA
>>   ;          C=Channel Number
>>   ;          R=RX Frequency
>>   ;          P=PL Frequency
>>   ;          A=Attributes
>>   ;
>>   ;Attributes : a=AM   b=LSB  f=FM  u=USB
>>   ;             l=Low Power   m=Medium Power   h=High Power
>>   ;             -=Minus TX Offset    s=Simplex     +=Plus TX Offset
>>   ;             t=TX PL on    r=RX PL on
>>   ;
>> 004=147.700,123.0,fshrt
>> 005=147.100,123.0,fshtt
>> 006=003.860,100.0,bshrt
>> init=003.980,100.0,bshrt ;Set to this on startup
>>
>>
>> This is just above the macro section below the functions-remote...
>>
>> At the end of the file I tried[memory]
>>
>> [memory]
>> 001=147.700,123.0,fshrt
>> 002=147.100,123.0,fshtt
>> 003=003.860,100.0,bshrt
>>
>>
>> None of these work, always says 'memory not found'
>>
>> Any ideas? I know I am getting close to making this work!
>>
>> 0=remote,1                             ; Retrieve Memory
>> 1=remote,2
>>
>> Mark KC7DMF
>>
>>
>>
>> ------------------------------
>>
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>>
>> ** --
> Sent from my Android phone with K-9 Mail. Please excuse my brevity and
> typing errors.html>
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From kj6ko at innercite.com  Sat Sep 21 15:55:44 2013
From: kj6ko at innercite.com (KJ6KO)
Date: Sat, 21 Sep 2013 08:55:44 -0700
Subject: [App_rpt-users] Fw: remote_timeout
Message-ID: <FCBCC104C44240E39F1957B8C0693403@KJ6KO>

Subject: remote_timeout


In rpt.conf, is the "remote_timeout=xxx"   for the RX timeout in case something on the system hangs up and will timeout after the time entered?  

In other words, the opposite of "totime=xxx" which is the TX timeout?

Thanks...



__________ Information from ESET NOD32 Antivirus, version of virus signature database 8827 (20130921) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From tim.sawyer at me.com  Sat Sep 21 16:12:06 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Sat, 21 Sep 2013 09:12:06 -0700
Subject: [App_rpt-users] remote_timeout
In-Reply-To: <FCBCC104C44240E39F1957B8C0693403@KJ6KO>
References: <FCBCC104C44240E39F1957B8C0693403@KJ6KO>
Message-ID: <5C5B8A47-BF54-45E3-B06D-772D2A3BA655@me.com>

http://ohnosec.org/drupal/search/node/remote_timeout
--
Tim
:wq

On Sep 21, 2013, at 8:55 AM, KJ6KO <kj6ko at innercite.com> wrote:

> Subject: remote_timeout
> 
> In rpt.conf, is the "remote_timeout=xxx"   for the RX timeout in case something on the system hangs up and will timeout after the time entered? 
>  
> In other words, the opposite of "totime=xxx" which is the TX timeout?
>  
> Thanks...
> 
> 
> 
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 8827 (20130921) __________
> 
> The message was checked by ESET NOD32 Antivirus.
> 
> http://www.eset.com
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From andy at ple.org  Sun Sep 22 00:22:34 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Sat, 21 Sep 2013 19:22:34 -0500
Subject: [App_rpt-users] Distorted audio
In-Reply-To: <1378951674.30020.YahooMailNeo@web141506.mail.bf1.yahoo.com>
References: <1378951674.30020.YahooMailNeo@web141506.mail.bf1.yahoo.com>
Message-ID: <ED7573AC-4187-4D89-901A-4FDA73E061F8@ple.org>

Though I should report back. I believe I fixed my distorted audio problem, and while doing so found a minor bug in app_rpt.c

Last week I needed to make some changes here and so I took the SSD I was using as the disk for the node out and replaced it with a real small 20GB hard disk. I know this is WAY too much, but that was the smaller one I had at home and so?

So I re-installed the node, uploaded the config file, EXACTLY the way they were and the distorted audio is gone. Since the ONLY thing I changed in my hard way setup, and the bug I found in app_rpt has/had nothing to do with audio delivery, the kernel is the same, ? logical deduction dictates if you eliminate everything else the remaining difference is most likely the culprit. In my case a perfectly functioning SSD. The only thing I can imagine is that maybe it was too fast, maybe it triggered some SATA3 (6GHb/s) bug, or maybe the MB I have is not really happy with the fast SATA or, ? in any case, since I switched to to the real hard disk instead of the small SSD (which I has no real use for anyway, because its only a 40GB) the node works flawlessly. 

Thanks to everyone offering help, it really helps getting all those ideas, and maybe the folks that have distorted audio as well can look at their HD setup, maybe its something similar.
--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 11, 2013, at 9:07 PM, Ryan Gross <n3ssl at yahoo.com> wrote:

> Ran into this problem a while ago and was found the URI interface was the problem with similar symptoms. Is this happening in duplex or simplex with audio ?
>  
> Ryan n3ssl
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From keith at goobie.org  Sun Sep 22 00:29:47 2013
From: keith at goobie.org (Keith Goobie)
Date: Sat, 21 Sep 2013 20:29:47 -0400
Subject: [App_rpt-users] Distorted audio
In-Reply-To: <ED7573AC-4187-4D89-901A-4FDA73E061F8@ple.org>
Message-ID: <CE63B23B.2378F%keith@goobie.org>

Andreas

Can you describe what the distortion sounded like.   We are running an SSD
here, but it has been good for us.

Keith


On 9/21/13 8:22 PM, "Andreas Pleschutznig" <andy at ple.org> wrote:

> Though I should report back. I believe I fixed my distorted audio problem, and
> while doing so found a minor bug in app_rpt.c
> 
> Last week I needed to make some changes here and so I took the SSD I was using
> as the disk for the node out and replaced it with a real small 20GB hard disk.
> I know this is WAY too much, but that was the smaller one I had at home and
> so?
> 
> So I re-installed the node, uploaded the config file, EXACTLY the way they
> were and the distorted audio is gone. Since the ONLY thing I changed in my
> hard way setup, and the bug I found in app_rpt has/had nothing to do with
> audio delivery, the kernel is the same, ? logical deduction dictates if you
> eliminate everything else the remaining difference is most likely the culprit.
> In my case a perfectly functioning SSD. The only thing I can imagine is that
> maybe it was too fast, maybe it triggered some SATA3 (6GHb/s) bug, or maybe
> the MB I have is not really happy with the fast SATA or, ? in any case, since
> I switched to to the real hard disk instead of the small SSD (which I has no
> real use for anyway, because its only a 40GB) the node works flawlessly.
> 
> Thanks to everyone offering help, it really helps getting all those ideas, and
> maybe the folks that have distorted audio as well can look at their HD setup,
> maybe its something similar.
> --
> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>  KA5PLE, Allstar: 29841, Echolonk: 884823
> 
> On Sep 11, 2013, at 9:07 PM, Ryan Gross <n3ssl at yahoo.com> wrote:
> 
>> Ran into this problem a while ago and was found the URI interface was the
>> problem with similar symptoms. Is this happening in duplex or simplex with
>> audio ?
>>  
>> Ryan n3ssl 
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

-- 
keith at goobie.org
Keith Goobie
Richmond Hill, ON, CANADA

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From andy at ple.org  Sun Sep 22 01:17:29 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Sat, 21 Sep 2013 20:17:29 -0500
Subject: [App_rpt-users] Distorted audio
In-Reply-To: <CE63B23B.2378F%keith@goobie.org>
References: <CE63B23B.2378F%keith@goobie.org>
Message-ID: <7F90F2F5-3AB4-4927-8828-C6E8017E4F1A@ple.org>

Hi Keith,

The way this sounded like was as if the audio is was hacked into 20ms pieces and every other piece was missing. I am pretty sure it is the combination of the specific MB and the SSD that was not OK together. That was the first problem I had with this SSD, and I am sure each of the devices on its own is perfectly fine, just not together. 


--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 21, 2013, at 7:29 PM, Keith Goobie <keith at goobie.org> wrote:

> Andreas
> 
> Can you describe what the distortion sounded like.   We are running an SSD here, but it has been good for us.
> 
> Keith
> 
> 
> On 9/21/13 8:22 PM, "Andreas Pleschutznig" <andy at ple.org> wrote:
> 
>> Though I should report back. I believe I fixed my distorted audio problem, and while doing so found a minor bug in app_rpt.c
>> 
>> Last week I needed to make some changes here and so I took the SSD I was using as the disk for the node out and replaced it with a real small 20GB hard disk. I know this is WAY too much, but that was the smaller one I had at home and so?
>> 
>> So I re-installed the node, uploaded the config file, EXACTLY the way they were and the distorted audio is gone. Since the ONLY thing I changed in my hard way setup, and the bug I found in app_rpt has/had nothing to do with audio delivery, the kernel is the same, ? logical deduction dictates if you eliminate everything else the remaining difference is most likely the culprit. In my case a perfectly functioning SSD. The only thing I can imagine is that maybe it was too fast, maybe it triggered some SATA3 (6GHb/s) bug, or maybe the MB I have is not really happy with the fast SATA or, ? in any case, since I switched to to the real hard disk instead of the small SSD (which I has no real use for anyway, because its only a 40GB) the node works flawlessly. 
>> 
>> Thanks to everyone offering help, it really helps getting all those ideas, and maybe the folks that have distorted audio as well can look at their HD setup, maybe its something similar.
>> --
>> Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
>>  KA5PLE, Allstar: 29841, Echolonk: 884823
>> 
>> On Sep 11, 2013, at 9:07 PM, Ryan Gross <n3ssl at yahoo.com> wrote:
>> 
>>> Ran into this problem a while ago and was found the URI interface was the problem with similar symptoms. Is this happening in duplex or simplex with audio ?
>>>  
>>> Ryan n3ssl 
>>> _______________________________________________
>>> App_rpt-users mailing list
>>> App_rpt-users at ohnosec.org
>>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>> 
>> 
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> -- 
> keith at goobie.org
> Keith Goobie
> Richmond Hill, ON, CANADA

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From N1XBM at amsat.org  Sun Sep 22 01:31:36 2013
From: N1XBM at amsat.org (Robert N. Newberry)
Date: Sat, 21 Sep 2013 21:31:36 -0400
Subject: [App_rpt-users] allstar sysop
Message-ID: <523E4878.1080507@gmail.com>

What is the consensus? I already have an allstar account. I see to
switch your account to a sysop it is non reversible. Should I create
another account for my node with my call sign N1XBM/R for sysop purposes?
-- 
Apparare Scientior
Paratus Communicare
-N1XBM


From buddy at brannan.name  Sun Sep 22 06:22:08 2013
From: buddy at brannan.name (Buddy Brannan)
Date: Sun, 22 Sep 2013 02:22:08 -0400
Subject: [App_rpt-users] allstar sysop
In-Reply-To: <523E4878.1080507@gmail.com>
References: <523E4878.1080507@gmail.com>
Message-ID: <EAB2F082-BBDE-43EC-9B00-7020F4CE349A@brannan.name>

Why? There?s no reason you can?t use your account for other purposes as well, i.e. dialing in on the phone or web transceiver. No need to create another one.
--
Buddy Brannan, KB5ELV - Erie, PA
Phone: (814) 860-3194 or 888-75-BUDDY



On Sep 21, 2013, at 9:31 PM, Robert N. Newberry <N1XBM at amsat.org> wrote:

> What is the consensus? I already have an allstar account. I see to
> switch your account to a sysop it is non reversible. Should I create
> another account for my node with my call sign N1XBM/R for sysop purposes?
> -- 
> Apparare Scientior
> Paratus Communicare
> -N1XBM
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



From zeppelin.1 at netzero.com  Sun Sep 22 10:09:27 2013
From: zeppelin.1 at netzero.com (zeppelin.1 at netzero.com)
Date: Sun, 22 Sep 2013 10:09:27 GMT
Subject: [App_rpt-users] AllStar Echolink
Message-ID: <20130922.060927.17552.0@webmail02.dca.untd.com>

Hello to all,
                Running two un-equal callsign and node number, AllStar in 1 P.C. How do get 2nd callsign Echolink node to operate.
 
                                                  73, Anton/ KC2RQR
                                                      IRLP node 4012            
                                                    AllStar Node 29103             
                                                      Echolink 361124

____________________________________________________________
One Weird Trick
Could add $1,000s to Your Social Security Checks! See if you Qualify&#8230
http://thirdpartyoffers.netzero.net/TGL3231/523ec22f40df5422f2030st03duc
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From k5tra at austin.rr.com  Sun Sep 22 15:09:45 2013
From: k5tra at austin.rr.com (tom)
Date: Sun, 22 Sep 2013 10:09:45 -0500
Subject: [App_rpt-users] AllStar Echolink
Message-ID: <001901ceb7a5$c5f6b650$51e422f0$@rr.com>

Anton,

You cannot run a second EchoLink node from the same IP address. Each
requires a separate IP and must use a different call (like KC2RQR-R and
KC2RQR-L). There is one way around the IP problem on a windows client by
using a proxy server for one of the nodes. I don't know how to run EchoLink
through a proxy from Allstar.

Tom / k5TRA

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From telesistant at hotmail.com  Sun Sep 22 15:33:08 2013
From: telesistant at hotmail.com (Jim Duuuude)
Date: Sun, 22 Sep 2013 08:33:08 -0700
Subject: [App_rpt-users] AllStar Echolink
In-Reply-To: <001901ceb7a5$c5f6b650$51e422f0$@rr.com>
References: <001901ceb7a5$c5f6b650$51e422f0$@rr.com>
Message-ID: <BLU180-W29F71DB2A49A4AD2C22E95B12C0@phx.gbl>

chan_echolink is MORE then capable of running as many echolink nodes as you want,
as long as your server has MULTIPLE public IP addresses. Since IP networking is MORE then
capable of allowing multiple IP addresses on a single interface, we certainly did not want to
limit users' ability to take advantage of this characteristic (unlike any other echolink implementation
that I am aware of).

Jim WB6NIL

From: k5tra at austin.rr.com
To: app_rpt-users at ohnosec.org
Date: Sun, 22 Sep 2013 10:09:45 -0500
Subject: [App_rpt-users] AllStar Echolink














Anton,

You cannot run a second EchoLink node from the same IP
address. Each requires a separate IP and must use a different call (like KC2RQR-R
and KC2RQR-L). There is one way around the IP problem on a windows client by
using a proxy server for one of the nodes. I don?t know how to run
EchoLink through a proxy from Allstar.

Tom / k5TRA







_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 		 	   		  
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From george at dyb.com  Sun Sep 22 18:31:56 2013
From: george at dyb.com (George Csahanin)
Date: Sun, 22 Sep 2013 13:31:56 -0500
Subject: [App_rpt-users] AllStar Echolink
References: <001901ceb7a5$c5f6b650$51e422f0$@rr.com>
	<BLU180-W29F71DB2A49A4AD2C22E95B12C0@phx.gbl>
Message-ID: <F1BEEB8BF21D4C1992BE0DFB199BED74@lintv.com>

Jim, how would one bind a second IP address to the different echolink nodes? I have zero intention of adding bad audio to my node, but real curious where that would get configured.

Thanks

GeorgeC
W2DB
2360
  ----- Original Message ----- 
  From: Jim Duuuude 
  To: tom ; app_rpt mailing list 
  Sent: Sunday, September 22, 2013 10:33 AM
  Subject: Re: [App_rpt-users] AllStar Echolink


  chan_echolink is MORE then capable of running as many echolink nodes as you want,
  as long as your server has MULTIPLE public IP addresses. Since IP networking is MORE then
  capable of allowing multiple IP addresses on a single interface, we certainly did not want to
  limit users' ability to take advantage of this characteristic (unlike any other echolink implementation
  that I am aware of).

  Jim WB6NIL



------------------------------------------------------------------------------
  From: k5tra at austin.rr.com
  To: app_rpt-users at ohnosec.org
  Date: Sun, 22 Sep 2013 10:09:45 -0500
  Subject: [App_rpt-users] AllStar Echolink


  Anton,

  You cannot run a second EchoLink node from the same IP address. Each requires a separate IP and must use a different call (like KC2RQR-R and KC2RQR-L). There is one way around the IP problem on a windows client by using a proxy server for one of the nodes. I don?t know how to run EchoLink through a proxy from Allstar.

  Tom / k5TRA


  _______________________________________________ App_rpt-users mailing list App_rpt-users at ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From N1XBM at amsat.org  Tue Sep 24 15:25:05 2013
From: N1XBM at amsat.org (Robert Newberry)
Date: Tue, 24 Sep 2013 11:25:05 -0400
Subject: [App_rpt-users] Getting there!
Message-ID: <CAM09xEbJG=+JC=x1HT62F+FaGU+q7_b6sg_cGf2-nZZzbS6dSw@mail.gmail.com>

OK so I got a node #, PC, repeater, URI, and I'm burning a ISO as we speak.

>From my rudimentary understanding it looks like a lot of the node
configuration is done online and the first time I update my PC I will need
to do a manual update.

So if I want to also use app_rpt as a controller I'm confused as to where I
set the important stuff, such as:

ID
Time out timer
Code to shut the transmitter off in the event something goes wrong.
PL tones encode/decode

I assume I must have to bring up a script and manually edit it? Any good
write ups in the internet about this?

I'm getting pretty excited about this. If I can get this one repeater on
the air. I have many more repeater that I want to link together for a
multicast system. I'm sure I'll have even more questions then.

Apparare Scientior
Paratus Communicare
-N1XBM
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From tim.sawyer at me.com  Tue Sep 24 16:15:33 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Tue, 24 Sep 2013 09:15:33 -0700
Subject: [App_rpt-users] Getting there!
In-Reply-To: <CAM09xEbJG=+JC=x1HT62F+FaGU+q7_b6sg_cGf2-nZZzbS6dSw@mail.gmail.com>
References: <CAM09xEbJG=+JC=x1HT62F+FaGU+q7_b6sg_cGf2-nZZzbS6dSw@mail.gmail.com>
Message-ID: <0015DCBD-F754-45E1-8847-0395EFE35040@me.com>

Some of that can happen on the portal. On the menu look at Node... Node configuration. What you can't set there must be set up by editing the configurations files. The two files where the things you mention are /etc/asterisk/rpt.conf and whatever channel driver you use (/etc/asterisk/simpleusb.conf or /etc/asterisk/usbradio.conf).
--
Tim
:wq

On Sep 24, 2013, at 8:25 AM, Robert Newberry <N1XBM at amsat.org> wrote:

> OK so I got a node #, PC, repeater, URI, and I'm burning a ISO as we speak.
> 
> From my rudimentary understanding it looks like a lot of the node configuration is done online and the first time I update my PC I will need to do a manual update.
> 
> So if I want to also use app_rpt as a controller I'm confused as to where I set the important stuff, such as:
> 
> ID
> Time out timer 
> Code to shut the transmitter off in the event something goes wrong.
> PL tones encode/decode
> 
> I assume I must have to bring up a script and manually edit it? Any good write ups in the internet about this?
> 
> I'm getting pretty excited about this. If I can get this one repeater on the air. I have many more repeater that I want to link together for a multicast system. I'm sure I'll have even more questions then.
> 
> Apparare Scientior 
> Paratus Communicare
> -N1XBM
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From N1XBM at amsat.org  Tue Sep 24 21:09:15 2013
From: N1XBM at amsat.org (Robert Newberry)
Date: Tue, 24 Sep 2013 17:09:15 -0400
Subject: [App_rpt-users] Getting there!
In-Reply-To: <0015DCBD-F754-45E1-8847-0395EFE35040@me.com>
References: <CAM09xEbJG=+JC=x1HT62F+FaGU+q7_b6sg_cGf2-nZZzbS6dSw@mail.gmail.com>
	<0015DCBD-F754-45E1-8847-0395EFE35040@me.com>
Message-ID: <CAM09xEZTyjky5ekSzo8trRjMvDD+rcgK-UqQy+mhtT8cKE-g0w@mail.gmail.com>

Thanks Tim,

I checked the allstar link site and checked the section you mentioned. I
also read up on simpleusb and usbradio on ohnosec.

I'm going to attempt a install tonight and see how it goes.

Apparare Scientior
Paratus Communicare
-N1XBM
On Sep 24, 2013 12:15 PM, "Tim Sawyer" <tim.sawyer at me.com> wrote:

> Some of that can happen on the portal. On the menu look at Node... Node
> configuration. What you can't set there must be set up by editing the
> configurations files. The two files where the things you mention are
> /etc/asterisk/rpt.conf and whatever channel driver you use
> (/etc/asterisk/simpleusb.conf or /etc/asterisk/usbradio.conf).
> --
> Tim
> :wq
>
> On Sep 24, 2013, at 8:25 AM, Robert Newberry <N1XBM at amsat.org> wrote:
>
> OK so I got a node #, PC, repeater, URI, and I'm burning a ISO as we speak.
>
> From my rudimentary understanding it looks like a lot of the node
> configuration is done online and the first time I update my PC I will need
> to do a manual update.
>
> So if I want to also use app_rpt as a controller I'm confused as to where
> I set the important stuff, such as:
>
> ID
> Time out timer
> Code to shut the transmitter off in the event something goes wrong.
> PL tones encode/decode
>
> I assume I must have to bring up a script and manually edit it? Any good
> write ups in the internet about this?
>
> I'm getting pretty excited about this. If I can get this one repeater on
> the air. I have many more repeater that I want to link together for a
> multicast system. I'm sure I'll have even more questions then.
>
> Apparare Scientior
> Paratus Communicare
> -N1XBM
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
>
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From jedscoot at yahoo.com  Tue Sep 24 22:09:35 2013
From: jedscoot at yahoo.com (Ian)
Date: Tue, 24 Sep 2013 23:09:35 +0100 (BST)
Subject: [App_rpt-users] Allmon+Lighttpd
Message-ID: <1380060575.40443.YahooMailNeo@web172006.mail.ir2.yahoo.com>

Hello,

I have Allmon running with Lighttpd on my PI and I have noticed while connected to a remote node via [ ssh ] to monitor, I see continual [ remote manager connect; remote manager disconnect ] prompts after a connect to the remote node from Allmon when logged in as admin.? There is no indication that the connection is connecting/disconnecting so I'm guessing that I just haven't quite set things up properly.
Any comments/suggestions most welcome.

73,
Ian..
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From keith at goobie.org  Tue Sep 24 22:11:50 2013
From: keith at goobie.org (Keith Goobie)
Date: Tue, 24 Sep 2013 18:11:50 -0400
Subject: [App_rpt-users] Allmon+Lighttpd
In-Reply-To: <1380060575.40443.YahooMailNeo@web172006.mail.ir2.yahoo.com>
Message-ID: <CE678666.23890%keith@goobie.org>

When allmon is active, it is constantly updating to the remote node.

You will see a bundle of traffic.

Keith


On 9/24/13 6:09 PM, "Ian" <jedscoot at yahoo.com> wrote:

> Hello,
> 
> I have Allmon running with Lighttpd on my PI and I have noticed while
> connected to a remote node via [ ssh ] to monitor, I see continual [ remote
> manager connect; remote manager disconnect ] prompts after a connect to the
> remote node from Allmon when logged in as admin.  There is no indication that
> the connection is connecting/disconnecting so I'm guessing that I just haven't
> quite set things up properly.
> Any comments/suggestions most welcome.
> 
> 73,
> Ian..
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

-- 
keith at goobie.org
Keith Goobie
Richmond Hill, ON, CANADA

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From N1XBM at amsat.org  Wed Sep 25 01:07:26 2013
From: N1XBM at amsat.org (Robert Newberry)
Date: Tue, 24 Sep 2013 21:07:26 -0400
Subject: [App_rpt-users] Getting there!
In-Reply-To: <CAM09xEZTyjky5ekSzo8trRjMvDD+rcgK-UqQy+mhtT8cKE-g0w@mail.gmail.com>
References: <CAM09xEbJG=+JC=x1HT62F+FaGU+q7_b6sg_cGf2-nZZzbS6dSw@mail.gmail.com>
	<0015DCBD-F754-45E1-8847-0395EFE35040@me.com>
	<CAM09xEZTyjky5ekSzo8trRjMvDD+rcgK-UqQy+mhtT8cKE-g0w@mail.gmail.com>
Message-ID: <CAM09xEZ0n-8=vDmoRJRugjmwBQ3uRQo5bWGJrzKuWRRPT7dJHA@mail.gmail.com>

Well software seems to installed without a hitch. The URI green light is
flashing and I get a "test login" prompt. I assume this is where I type
root for login and my password I created.

So I guess next I need the radio connected and the levels set.

Where can I read more about adding lines to definitions? I'm reading up on
status reporting and adding two lines to /etc/asterisk/rpt.conf:

Is this as simple as being logged in and entering that line and adding the
info to the bottom of the file?

Thank you

Apparare Scientior
Paratus Communicare
-N1XBM
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From N1XBM at amsat.org  Wed Sep 25 02:15:33 2013
From: N1XBM at amsat.org (Robert N. Newberry)
Date: Tue, 24 Sep 2013 22:15:33 -0400
Subject: [App_rpt-users] some more progress
Message-ID: <52424745.6070902@gmail.com>

So I was able to get manual download and now my login screen shows
N1XBM/R. I also now show up on allstar link. I tried connecting to my
node (radio not connected yet) with the web transceiver I hit connect
and nothing. I tried hitting the number keys and nothing. I also tried
an automatic download of my server config and that did not appear to
work. It said my server should reboot when done and I did not get a reboot.

So I guess I am making some progress here. I need to figure out the web
transceiver and why I can not download my server config from the website.


-- 
Apparare Scientior
Paratus Communicare
-N1XBM


From tim.sawyer at me.com  Wed Sep 25 02:53:00 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Tue, 24 Sep 2013 19:53:00 -0700
Subject: [App_rpt-users] Allmon+Lighttpd
In-Reply-To: <1380060575.40443.YahooMailNeo@web172006.mail.ir2.yahoo.com>
References: <1380060575.40443.YahooMailNeo@web172006.mail.ir2.yahoo.com>
Message-ID: <E4F8AA32-231F-40E3-8E60-62581C88347D@me.com>

Allmon does continuously login to the manager to get node's status and then logout. The messages you see are normal Allmon operation.
--
Tim
:wq

On Sep 24, 2013, at 3:09 PM, Ian <jedscoot at yahoo.com> wrote:

> Hello,
> 
> I have Allmon running with Lighttpd on my PI and I have noticed while connected to a remote node via [ ssh ] to monitor, I see continual [ remote manager connect; remote manager disconnect ] prompts after a connect to the remote node from Allmon when logged in as admin.  There is no indication that the connection is connecting/disconnecting so I'm guessing that I just haven't quite set things up properly.
> Any comments/suggestions most welcome.
> 
> 73,
> Ian..
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From tim.sawyer at me.com  Wed Sep 25 03:02:26 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Tue, 24 Sep 2013 20:02:26 -0700
Subject: [App_rpt-users] some more progress
In-Reply-To: <52424745.6070902@gmail.com>
References: <52424745.6070902@gmail.com>
Message-ID: <B40814EE-A93C-477F-BF33-126D76F002A3@me.com>

Make sure you can connect to some other node than your own to test the webtransceiver. It's kinda buggy and may not be working for you. 

If that's ok what you may have is poor nat-tranversal... meaning the webtransceiver has to connect to your public IP and thus traverse out of your router and then back into your router and to the node. Some routers don't do that well. 

Also make sure you have 4567, or whichever IAX port you are using forwarded to your node. 

Have someone connect to your node, just ask anyone here will be glad to try it. 

Remember, a node has to be on line for a long while before it learns all the IP address of the other nodes and before the other nodes learn about your IP. Sometimes that can take up to a half an hour. 
--
Tim
:wq

On Sep 24, 2013, at 7:15 PM, Robert N. Newberry <N1XBM at amsat.org> wrote:

> So I was able to get manual download and now my login screen shows
> N1XBM/R. I also now show up on allstar link. I tried connecting to my
> node (radio not connected yet) with the web transceiver I hit connect
> and nothing. I tried hitting the number keys and nothing. I also tried
> an automatic download of my server config and that did not appear to
> work. It said my server should reboot when done and I did not get a reboot.
> 
> So I guess I am making some progress here. I need to figure out the web
> transceiver and why I can not download my server config from the website.
> 
> 
> -- 
> Apparare Scientior
> Paratus Communicare
> -N1XBM
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users



From petem001 at hotmail.com  Wed Sep 25 03:04:45 2013
From: petem001 at hotmail.com (Pierre Martel)
Date: Tue, 24 Sep 2013 23:04:45 -0400
Subject: [App_rpt-users] radio tune settings
In-Reply-To: <52424745.6070902@gmail.com>
References: <52424745.6070902@gmail.com>
Message-ID: <BAY167-DS16B0CA6526D81DCC3CF84E972F0@phx.gbl>

I have a 2 node setup on repeater, one for vhf and one for vhf.

I run Acid and want to run the setup on a 64 bit machine. I have seen that 
Xipar will run on 64 bit arch.

If I transfert my node to a xipar machine will have to retune my levels with 
radio tune menu or will I be able to transfert the info in the 
usbradio_tune_usbxxxxx.conf with xipar?




From cypresstower at yahoo.com  Wed Sep 25 03:32:48 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Tue, 24 Sep 2013 20:32:48 -0700 (PDT)
Subject: [App_rpt-users] radio tune settings
In-Reply-To: <BAY167-DS16B0CA6526D81DCC3CF84E972F0@phx.gbl>
References: <52424745.6070902@gmail.com>
	<BAY167-DS16B0CA6526D81DCC3CF84E972F0@phx.gbl>
Message-ID: <1380079968.40400.YahooMailNeo@web163604.mail.gq1.yahoo.com>

May not be of any help but, did you uncomment the displayconnects = yes in the /etc/asterisk/manager.conf and change it to no???
JK


From: Pierre Martel <petem001 at hotmail.com>
To: app_rpt-users at ohnosec.org 
Sent: Tuesday, September 24, 2013 11:04 PM
Subject: [App_rpt-users] radio tune settings


I have a 2 node setup on repeater, one for vhf and one for vhf.

I run Acid and want to run the setup on a 64 bit machine. I have seen that 
Xipar will run on 64 bit arch.

If I transfert my node to a xipar machine will have to retune my levels with 
radio tune menu or will I be able to transfert the info in the 
usbradio_tune_usbxxxxx.conf with xipar?


_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From cypresstower at yahoo.com  Wed Sep 25 03:38:41 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Tue, 24 Sep 2013 20:38:41 -0700 (PDT)
Subject: [App_rpt-users] some more progress
In-Reply-To: <52424745.6070902@gmail.com>
References: <52424745.6070902@gmail.com>
Message-ID: <1380080321.19098.YahooMailNeo@web163603.mail.gq1.yahoo.com>

The nodesetup.sh script may help???? try it from your root prompt
[root at N1XBM ~]#nodesetup.sh??
follow the prompts
From: Robert N. Newberry <N1XBM at amsat.org>
To: "app_rpt-users at ohnosec.org" <app_rpt-users at ohnosec.org> 
Sent: Tuesday, September 24, 2013 10:15 PM
Subject: [App_rpt-users] some more progress


So I was able to get manual download and now my login screen shows
N1XBM/R. I also now show up on allstar link. I tried connecting to my
node (radio not connected yet) with the web transceiver I hit connect
and nothing. I tried hitting the number keys and nothing. I also tried
an automatic download of my server config and that did not appear to
work. It said my server should reboot when done and I did not get a reboot.

So I guess I am making some progress here. I need to figure out the web
transceiver and why I can not download my server config from the website.


-- 
Apparare Scientior
Paratus Communicare
-N1XBM
_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From cypresstower at yahoo.com  Wed Sep 25 03:39:56 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Tue, 24 Sep 2013 20:39:56 -0700 (PDT)
Subject: [App_rpt-users] Allmon+Lighttpd
In-Reply-To: <1380060575.40443.YahooMailNeo@web172006.mail.ir2.yahoo.com>
References: <1380060575.40443.YahooMailNeo@web172006.mail.ir2.yahoo.com>
Message-ID: <1380080396.84057.YahooMailNeo@web163606.mail.gq1.yahoo.com>

May not be of any help but, did you uncomment the displayconnects = yes in the /etc/asterisk/manager.conf and change it to no???
JK


From: Ian <jedscoot at yahoo.com>
To: "App_rpt-users at ohnosec.org" <App_rpt-users at ohnosec.org> 
Sent: Tuesday, September 24, 2013 6:09 PM
Subject: [App_rpt-users] Allmon+Lighttpd



Hello,

I have Allmon running with Lighttpd on my PI and I have noticed while connected to a remote node via [ ssh ] to monitor, I see continual [ remote manager connect; remote manager disconnect ] prompts after a connect to the remote node from Allmon when logged in as admin.? There is no indication that the connection is connecting/disconnecting so I'm guessing that I just haven't quite set things up properly.
Any comments/suggestions most welcome.

73,
Ian..
_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From cypresstower at yahoo.com  Wed Sep 25 03:45:24 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Tue, 24 Sep 2013 20:45:24 -0700 (PDT)
Subject: [App_rpt-users] radio tune settings
In-Reply-To: <BAY167-DS16B0CA6526D81DCC3CF84E972F0@phx.gbl>
References: <52424745.6070902@gmail.com>
	<BAY167-DS16B0CA6526D81DCC3CF84E972F0@phx.gbl>
Message-ID: <1380080724.17129.YahooMailNeo@web163606.mail.gq1.yahoo.com>

Yes!! but?the radio tune settings are set in the CLI 
also get the docs on the wan page for the filter settings, you'll love it.
Don't give up on acid, it has advantages like allmon.


From: Pierre Martel <petem001 at hotmail.com>
To: app_rpt-users at ohnosec.org 
Sent: Tuesday, September 24, 2013 11:04 PM
Subject: [App_rpt-users] radio tune settings


I have a 2 node setup on repeater, one for vhf and one for vhf.

I run Acid and want to run the setup on a 64 bit machine. I have seen that 
Xipar will run on 64 bit arch.

If I transfert my node to a xipar machine will have to retune my levels with 
radio tune menu or will I be able to transfert the info in the 
usbradio_tune_usbxxxxx.conf with xipar?


_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From n3fe at repeater.net  Wed Sep 25 04:40:46 2013
From: n3fe at repeater.net (Corey Dean)
Date: Wed, 25 Sep 2013 00:40:46 -0400
Subject: [App_rpt-users] radio tune settings
In-Reply-To: <BAY167-DS16B0CA6526D81DCC3CF84E972F0@phx.gbl>
References: <52424745.6070902@gmail.com>
	<BAY167-DS16B0CA6526D81DCC3CF84E972F0@phx.gbl>
Message-ID: <4BCC91CBCFD66C4489B4BD3233140C3E048596B6D2CD@exchange.mail.repeater.net>

Chances are you will want to retune if you will be using the wider codecs.  

Corey  N3FE

-----Original Message-----
From: app_rpt-users-bounces at ohnosec.org [mailto:app_rpt-users-bounces at ohnosec.org] On Behalf Of Pierre Martel
Sent: Tuesday, September 24, 2013 11:05 PM
To: app_rpt-users at ohnosec.org
Subject: [App_rpt-users] radio tune settings

I have a 2 node setup on repeater, one for vhf and one for vhf.

I run Acid and want to run the setup on a 64 bit machine. I have seen that Xipar will run on 64 bit arch.

If I transfert my node to a xipar machine will have to retune my levels with radio tune menu or will I be able to transfert the info in the usbradio_tune_usbxxxxx.conf with xipar?


_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

--
This message was scanned and is believed to be clean.
Click here to report this message as spam. 
http://simba.repeater.net/cgi-bin/learn-msg.cgi?id=3F8702228.A74B8




From jedscoot at yahoo.com  Wed Sep 25 09:54:13 2013
From: jedscoot at yahoo.com (Ian)
Date: Wed, 25 Sep 2013 10:54:13 +0100 (BST)
Subject: [App_rpt-users] Allmon+Lighttpd
In-Reply-To: <1380080396.84057.YahooMailNeo@web163606.mail.gq1.yahoo.com>
References: <1380060575.40443.YahooMailNeo@web172006.mail.ir2.yahoo.com>
	<1380080396.84057.YahooMailNeo@web163606.mail.gq1.yahoo.com>
Message-ID: <1380102853.83882.YahooMailNeo@web172004.mail.ir2.yahoo.com>

Thanks for the comments;? it was indeed the [ displayconnects = yes ] that I needed to change to [ no ] on the remote node.

73,
Ian..




________________________________
 From: Johnny Keeker <cypresstower at yahoo.com>
To: Ian <jedscoot at yahoo.com>; "App_rpt-users at ohnosec.org" <App_rpt-users at ohnosec.org> 
Sent: Wednesday, 25 September 2013, 4:39
Subject: Re: [App_rpt-users] Allmon+Lighttpd
 


May not be of any help but, did you uncomment the displayconnects = yes in the /etc/asterisk/manager.conf and change it to no???
JK


From: Ian <jedscoot at yahoo.com>
To: "App_rpt-users at ohnosec.org" <App_rpt-users at ohnosec.org> 
Sent: Tuesday, September 24, 2013 6:09 PM
Subject: [App_rpt-users] Allmon+Lighttpd



Hello,

I have Allmon running with Lighttpd on my PI and I have noticed while connected to a remote node via [ ssh ] to monitor, I see continual [ remote manager connect; remote manager disconnect ] prompts after a connect to the remote node from Allmon when logged in as admin.? There is no indication that the connection is connecting/disconnecting so I'm guessing that I just haven't quite set things up properly.
Any comments/suggestions most welcome.

73,
Ian..
_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From vk3ant at optusnet.com.au  Thu Sep 26 04:45:35 2013
From: vk3ant at optusnet.com.au (R.K.)
Date: Thu, 26 Sep 2013 14:45:35 +1000
Subject: [App_rpt-users] Help please with Beagle/LOX combination.
Message-ID: <4553E5FE03BA422CBB0648CA35ED45A2@RKPC>


Greetings,

Wonder if there is someone out there who would be kind enough to ? take me by the hand ?, figuratively speaking, and guide me one step at a time, through the process of getting my Beagle/LOX combo up and running.

I have had this thing for months now and I have had some help from one fellow, but for whatever reason I just can?t get the numbers to fall in to place, so now I want to go back to the start and have another go.

Where I am currently at is, I am able to log on to another Node from my Beagle, but I believe my transmitted audio is very rough and difficult to understand, but then the next issue is, I am unable to hear the person I am connected to.

I have been following the rantings on this forum for a while now and been applying bit and pieces that I believe relevant to my situation, but I still come up short of a perfectly working combination.

I have no need for much of what many of you guys out there are doing with your gear, I am retired, wife and I do a bit of driving around the country, and all I want to do with my gear is that when we set up camp for the evening somewhere is drag the Beagle/LOX out from the back of the car, set it up and just talk. Nothing fancy.

Anyway, if any of you out there are willing to take me under your wing and guide me through the process of getting this thing working then it would be very much appreciated, please contact me off list at vk3ant at arrl.net 

Thanks everyone.

Ron
VK3ANT
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From ke2n at cs.com  Thu Sep 26 08:53:47 2013
From: ke2n at cs.com (Ken)
Date: Thu, 26 Sep 2013 04:53:47 -0400
Subject: [App_rpt-users] 64 bit
Message-ID: <006b01ceba95$e9c5a260$bd50e720$@cs.com>



Basically all computers you buy today are 64 bit architecture.  If you mean
64 bit operating system (64 bit version of CENTOS) app-rpt will probably run
on that.  A couple of years ago, as an experiment I installed a 64 bit
system, then ran the ACID install script and - while it made various warning
messages during the compile -  it ran just fine.   Over the years I have
seen where several other people have done the same.

 

73

 

Ken

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From petem001 at hotmail.com  Thu Sep 26 10:47:21 2013
From: petem001 at hotmail.com (pete M)
Date: Thu, 26 Sep 2013 06:47:21 -0400
Subject: [App_rpt-users] =?iso-8859-1?q?Re=A0=3A_64_bit?=
Message-ID: <BAY405-EAS85289EEDE78C79768D1FAD97280@phx.gbl>

That did not work here and I dont know why. But as soon that I switched to centos 6 All my device worked perfectly. So I need centos 6 . But the acid instalk script fail all the time. And I dont have the time to modify it. But at the same time the xipar install worked on first try. This will also give mw thw chance to try the voting receiver stuff for cheap cause I have some thin client and usb dongle that I will connect to some gm300 that have the output blown but receive perfectly.

--- Message initial ---

De : "Ken" <ke2n at cs.com>
Envoy? : 26 septembre 2013 04:54
A : petem001 at hotmail.com
Cc: app_rpt-users at ohnosec.org
Objet : 64 bit



Basically all computers you buy today are 64 bit architecture.  If you mean
64 bit operating system (64 bit version of CENTOS) app-rpt will probably run
on that.  A couple of years ago, as an experiment I installed a 64 bit
system, then ran the ACID install script and - while it made various warning
messages during the compile -  it ran just fine.   Over the years I have
seen where several other people have done the same.



73



Ken

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From cypresstower at yahoo.com  Thu Sep 26 16:36:19 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Thu, 26 Sep 2013 09:36:19 -0700 (PDT)
Subject: [App_rpt-users] 64 bit
In-Reply-To: <006b01ceba95$e9c5a260$bd50e720$@cs.com>
References: <006b01ceba95$e9c5a260$bd50e720$@cs.com>
Message-ID: <1380213379.20029.YahooMailNeo@web163603.mail.gq1.yahoo.com>

What about the ACID Centos6 on the SVN site.? Has anyone else tried it beside me.
JK

From: Ken <ke2n at cs.com>
To: petem001 at hotmail.com 
Cc: app_rpt-users at ohnosec.org 
Sent: Thursday, September 26, 2013 4:53 AM
Subject: [App_rpt-users] 64 bit



 
Basically all computers you buy today are 64 bit architecture.? If you mean 64 bit operating system (64 bit version of CENTOS) app-rpt will probably run on that.? A couple of years ago, as an experiment I installed a 64 bit system, then ran the ACID install script and ? while it made various warning messages during the compile - ?it ran just fine.?? Over the years I have seen where several other people have done the same.
?
73
?
Ken
_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From N1XBM at amsat.org  Fri Sep 27 13:42:33 2013
From: N1XBM at amsat.org (Robert Newberry)
Date: Fri, 27 Sep 2013 09:42:33 -0400
Subject: [App_rpt-users] Control opp codes
In-Reply-To: <CAM09xEbH76xH=u9kmQH+Z_qoMJ+rytqqsDw2=qxi+GUvZAg8MA@mail.gmail.com>
References: <CAM09xEZynXZC9hu3U8ePNDkR5GPNbAjnDxNhsK4jY-0qzkO=xw@mail.gmail.com>
	<CAM09xEbH76xH=u9kmQH+Z_qoMJ+rytqqsDw2=qxi+GUvZAg8MA@mail.gmail.com>
Message-ID: <CAM09xEao=T3Rw43S2KjTb45w8J4DDr3N9E0SiHF7wXbFrF4oRQ@mail.gmail.com>

Well I'm making headway here. Now I'm confused again. I was looking at a
way to have the URI stop asserting the PTT line after hearing a certain
DTMF.

I see cop, 3 ;system disable.

Does this mean DTMF 3 will disable the system? I noticed on ohnosec the
rpt.conf.sample it says change these to something other than listed below.

Also what exactly happens on a system disable?

Thank you

Apparare Scientior
Paratus Communicare
-N1XBM
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From tim.sawyer at me.com  Fri Sep 27 14:32:09 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Fri, 27 Sep 2013 07:32:09 -0700
Subject: [App_rpt-users] Control opp codes
In-Reply-To: <CAM09xEao=T3Rw43S2KjTb45w8J4DDr3N9E0SiHF7wXbFrF4oRQ@mail.gmail.com>
References: <CAM09xEZynXZC9hu3U8ePNDkR5GPNbAjnDxNhsK4jY-0qzkO=xw@mail.gmail.com>
	<CAM09xEbH76xH=u9kmQH+Z_qoMJ+rytqqsDw2=qxi+GUvZAg8MA@mail.gmail.com>
	<CAM09xEao=T3Rw43S2KjTb45w8J4DDr3N9E0SiHF7wXbFrF4oRQ@mail.gmail.com>
Message-ID: <FE9470AE-C039-4EFA-B48A-FA6B6AD379C6@me.com>

System disable turns off the transmitter. 

To use cop 3 you would add it your [functions] section in /etc/asterisk/rpt.conf for example:

950 = cop,3    ; Disable system
951 = cop,2    ; Enable system

Now *950 will turn off the system and *951 will turn it on.
--
Tim
:wq

On Sep 27, 2013, at 6:42 AM, Robert Newberry <N1XBM at amsat.org> wrote:

> Well I'm making headway here. Now I'm confused again. I was looking at a way to have the URI stop asserting the PTT line after hearing a certain DTMF.
> 
> I see cop, 3 ;system disable.
> 
> Does this mean DTMF 3 will disable the system? I noticed on ohnosec the rpt.conf.sample it says change these to something other than listed below.
> 
> Also what exactly happens on a system disable?
> 
> Thank you
> 
> Apparare Scientior 
> Paratus Communicare
> -N1XBM
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From N1XBM at amsat.org  Fri Sep 27 14:45:03 2013
From: N1XBM at amsat.org (Robert Newberry)
Date: Fri, 27 Sep 2013 10:45:03 -0400
Subject: [App_rpt-users] Control opp codes
In-Reply-To: <FE9470AE-C039-4EFA-B48A-FA6B6AD379C6@me.com>
References: <CAM09xEZynXZC9hu3U8ePNDkR5GPNbAjnDxNhsK4jY-0qzkO=xw@mail.gmail.com>
	<CAM09xEbH76xH=u9kmQH+Z_qoMJ+rytqqsDw2=qxi+GUvZAg8MA@mail.gmail.com>
	<CAM09xEao=T3Rw43S2KjTb45w8J4DDr3N9E0SiHF7wXbFrF4oRQ@mail.gmail.com>
	<FE9470AE-C039-4EFA-B48A-FA6B6AD379C6@me.com>
Message-ID: <CAM09xEYorCNp7F9JKMVdq_+7+juqQq3sy+FP6YFxYi58FQ55aQ@mail.gmail.com>

OK, I see in the copy of rpt.conf.sample right now

 ;92=cop,3     ;system disable

So in theory I would want to change 92 to something I would prefer like
your using 950 and 951.

I think I have that right!

Thanks


On Fri, Sep 27, 2013 at 10:32 AM, Tim Sawyer <tim.sawyer at me.com> wrote:

> System disable turns off the transmitter.
>
> To use cop 3 you would add it your [functions] section in
> /etc/asterisk/rpt.conf for example:
>
> 950 = cop,3    ; Disable system
> 951 = cop,2    ; Enable system
>
> Now *950 will turn off the system and *951 will turn it on.
> --
> Tim
> :wq
>
> On Sep 27, 2013, at 6:42 AM, Robert Newberry <N1XBM at amsat.org> wrote:
>
> Well I'm making headway here. Now I'm confused again. I was looking at a
> way to have the URI stop asserting the PTT line after hearing a certain
> DTMF.
>
> I see cop, 3 ;system disable.
>
> Does this mean DTMF 3 will disable the system? I noticed on ohnosec the
> rpt.conf.sample it says change these to something other than listed below.
>
> Also what exactly happens on a system disable?
>
> Thank you
>
> Apparare Scientior
> Paratus Communicare
> -N1XBM
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
>
>
>
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From tim.sawyer at me.com  Fri Sep 27 16:41:23 2013
From: tim.sawyer at me.com (Tim Sawyer)
Date: Fri, 27 Sep 2013 09:41:23 -0700
Subject: [App_rpt-users] Control opp codes
In-Reply-To: <CAM09xEYorCNp7F9JKMVdq_+7+juqQq3sy+FP6YFxYi58FQ55aQ@mail.gmail.com>
References: <CAM09xEZynXZC9hu3U8ePNDkR5GPNbAjnDxNhsK4jY-0qzkO=xw@mail.gmail.com>
	<CAM09xEbH76xH=u9kmQH+Z_qoMJ+rytqqsDw2=qxi+GUvZAg8MA@mail.gmail.com>
	<CAM09xEao=T3Rw43S2KjTb45w8J4DDr3N9E0SiHF7wXbFrF4oRQ@mail.gmail.com>
	<FE9470AE-C039-4EFA-B48A-FA6B6AD379C6@me.com>
	<CAM09xEYorCNp7F9JKMVdq_+7+juqQq3sy+FP6YFxYi58FQ55aQ@mail.gmail.com>
Message-ID: <CDD33542-A2F1-47C1-9F6E-41AE73B0F79E@me.com>

No, you can leave it as is. That way *92 is your disable command. 950 and 951 were just example. 
--
Tim
:wq

On Sep 27, 2013, at 7:45 AM, Robert Newberry <N1XBM at amsat.org> wrote:

> OK, I see in the copy of rpt.conf.sample right now
> 
>  ;92=cop,3     ;system disable
> 
> So in theory I would want to change 92 to something I would prefer like your using 950 and 951.
> 
> I think I have that right!
> 
> Thanks
> 
> 
> On Fri, Sep 27, 2013 at 10:32 AM, Tim Sawyer <tim.sawyer at me.com> wrote:
> System disable turns off the transmitter. 
> 
> To use cop 3 you would add it your [functions] section in /etc/asterisk/rpt.conf for example:
> 
> 950 = cop,3    ; Disable system
> 951 = cop,2    ; Enable system
> 
> Now *950 will turn off the system and *951 will turn it on.
> --
> Tim
> :wq
> 
> On Sep 27, 2013, at 6:42 AM, Robert Newberry <N1XBM at amsat.org> wrote:
> 
>> Well I'm making headway here. Now I'm confused again. I was looking at a way to have the URI stop asserting the PTT line after hearing a certain DTMF.
>> 
>> I see cop, 3 ;system disable.
>> 
>> Does this mean DTMF 3 will disable the system? I noticed on ohnosec the rpt.conf.sample it says change these to something other than listed below.
>> 
>> Also what exactly happens on a system disable?
>> 
>> Thank you
>> 
>> Apparare Scientior 
>> Paratus Communicare
>> -N1XBM
>> 
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at ohnosec.org
>> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From N1XBM at amsat.org  Sat Sep 28 02:24:50 2013
From: N1XBM at amsat.org (Robert N. Newberry)
Date: Fri, 27 Sep 2013 22:24:50 -0400
Subject: [App_rpt-users] uncommnet
Message-ID: <52463DF2.7030601@gmail.com>

So I've gotten into the usbradio.conf file and made some changes. I've
also looked at the rpt.conf I was looking at the status reporting
section and depending on if I'm running acid or limey. I need to
uncomment. I did a startpage search and found reference to removing #
but I do not see that in the file. What am I missing?
-- 
Apparare Scientior
Paratus Communicare
-N1XBM


From wolthuis at gmail.com  Sat Sep 28 02:38:44 2013
From: wolthuis at gmail.com (Michael Wolthuis)
Date: Fri, 27 Sep 2013 22:38:44 -0400
Subject: [App_rpt-users] COR on MotoTrbo XPR4350?
In-Reply-To: <52463DF2.7030601@gmail.com>
Message-ID: <CE6BB932.1B765%wolthuis@gmail.com>

I understand the concerns, but I am interested in trying..  I want to hook
a MotoTrbo XPR4350 to my beagle board.

Does anyone know which pin on the XPR is COR?  Is there a CTCSS pin?

Thanks for any advice,
Mike
kb8zgl




From wl7lp at yahoo.com  Sat Sep 28 03:08:16 2013
From: wl7lp at yahoo.com (Russ WL7LP)
Date: Fri, 27 Sep 2013 20:08:16 -0700 (PDT)
Subject: [App_rpt-users] which file
Message-ID: <1380337696.20158.YahooMailNeo@web163104.mail.bf1.yahoo.com>

I bought a URI for my Allstar node. I am having problem with it at this time.
problem is when I plug the URI into a usb port it keys up everything that is connected to me
or I'm connected to. it does this with or without a radio plugged into the URI. to unkey the node
I have to unplug the URI from the usb port. and it don't matter which usb port it's plug into.

which file should the URI use? as I see 2 possible files. one called usbradio.conf and the other
simpleusb.conf.? 

?
73 Russ WL7LP
AllStar 29332, 29265
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From n5zua at earthlink.net  Sat Sep 28 05:13:36 2013
From: n5zua at earthlink.net (Steve Agee)
Date: Sat, 28 Sep 2013 00:13:36 -0500
Subject: [App_rpt-users] uncommnet
References: <52463DF2.7030601@gmail.com>
Message-ID: <9836C8E65CBC4D86A254F53A9514EC4D@steveea3dc3d27>

You want to remove the semicolon, which is the comment character used in 
these types of files. The # is used for comments in other areas, such as in 
a crontab.

N5ZUA

----- Original Message ----- 
From: "Robert N. Newberry" <N1XBM at amsat.org>
To: <app_rpt-users at ohnosec.org>
Sent: Friday, September 27, 2013 9:24 PM
Subject: [App_rpt-users] uncommnet


> So I've gotten into the usbradio.conf file and made some changes. I've
> also looked at the rpt.conf I was looking at the status reporting
> section and depending on if I'm running acid or limey. I need to
> uncomment. I did a startpage search and found reference to removing #
> but I do not see that in the file. What am I missing?
> -- 
> Apparare Scientior
> Paratus Communicare
> -N1XBM
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users 



From keith at handscombe.co.uk  Sat Sep 28 16:14:33 2013
From: keith at handscombe.co.uk (keith)
Date: Sat, 28 Sep 2013 17:14:33 +0100
Subject: [App_rpt-users] Mute
Message-ID: <000001cebc65$d1ff4790$75fdd6b0$@co.uk>

Evening all,

 

It's been a long time since I am sure I did this on my first node build. In
the morning I connect to the Winsystem and my node then gets then connects
and disconnects while this net is on. I would prefer to only get the first
point that my node connect to and then disconnects from and have no other
node reports in-between the actual net. I have removed the Echo link
connection broadcasts to reduce but would like to reduce the All-Star
connection broadcasts please.

 

Whomever helps thanks

 

   Kind Regards

 

   Keith Handscombe     

 

154e886

 

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From ars.w5omr at gmail.com  Sat Sep 28 17:01:04 2013
From: ars.w5omr at gmail.com (Geoff Edmonson)
Date: Sat, 28 Sep 2013 12:01:04 -0500
Subject: [App_rpt-users] Mute
Message-ID: <yvs6mpmcg2hinmbmyyoe9h8t.1380387664772@email.android.com>


Did any one else understand this?



keith <keith at handscombe.co.uk> wrote:

>_______________________________________________
>App_rpt-users mailing list
>App_rpt-users at ohnosec.org
>http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From jrorke at cogeco.ca  Sat Sep 28 17:14:27 2013
From: jrorke at cogeco.ca (REDBUTTON_CTRL)
Date: Sat, 28 Sep 2013 13:14:27 -0400
Subject: [App_rpt-users] Mute
In-Reply-To: <000001cebc65$d1ff4790$75fdd6b0$@co.uk>
References: <000001cebc65$d1ff4790$75fdd6b0$@co.uk>
Message-ID: <52470E73.1090205@cogeco.ca>

If you are wanting to quiet the connect and disconnect messages on you 
node, then put these 2 lines in you rpt.conf file:

holdofftelem=1    ; polite telemetry - waits until the channel is not 
busy to send telemetry locally
telemdefault=2    ; timed local telemetry - announces connect/ 
disconnect messages after a local is executed then turns off all voice 
announcements after 5 minutes.


The telemdefault=2 command allows local announcements after a local 
command is executed. After a while with no activity locally the 
announcements wont come on. If you issue a connect then it announces 
locally on you node. But after a 5 minutes or so the announcements will 
quiet down and stay off unless you issuue a command locally again and 
that turns the announcements back on for 5 min.

This works great when you are on a linked system system where nodes are 
constantly connecting and disconnecting.

Hope this is what you are looking for.

Jon VA3RQ

On 9/28/2013 12:14 PM, keith wrote:
>
> Evening all,
>
> It's been a long time since I am sure I did this on my first node 
> build. In the morning I connect to the Winsystem and my node then gets 
> then connects and disconnects while this net is on. I would prefer to 
> only get the first point that my node connect to and then disconnects 
> from and have no other node reports in-between the actual net. I have 
> removed the Echo link connection broadcasts to reduce but would like 
> to reduce the All-Star connection broadcasts please.
>
> Whomever helps thanks
>
>    Kind Regards
>
>    Keith Handscombe
>
> 154e886
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From telesistant at hotmail.com  Sat Sep 28 17:55:44 2013
From: telesistant at hotmail.com (Jim Duuuude)
Date: Sat, 28 Sep 2013 10:55:44 -0700
Subject: [App_rpt-users] Simulcast Coverage Plots (For Free)
Message-ID: <BLU180-W7828031C91F1B47D5628B8B12A0@phx.gbl>

Amongst many other things, I have been working on adapting the AMAZING
open-source radio transmitter coverage program SPLAT to be able to generate
a (hopefully) meaningful coverage plot (graphical) including simulcast transmitter
interactions.

I have made the necessary changes/additions and now wish to offer this ability
to all Amateurs that are interested at no charge. 

Unfortunately, this process is EXTREMELY CPU-intensive, and I *just* dont have
a server available that has enough processing "oomph" to accomplish this in a
reasonable manner.

For example, the private server that I have been using for develoment is a single
(quad-core) Xeon (older one), and it takes OVER 20 MINUTES to generate a 2
transmitter plot!!

Does anyone have some server "space" on a REALLY fast system that could be
"donated" for this purpose? We would *ALL* appreciate this VERY much!! :-)

I will shortly attach some screenshots of the program as it currently exists. Since I
really dont want anyone "test driving" the software on the current server, the URL
was removed from the graphics. :-)

I should have a copy of the sources of this 'modified' splat up on SVN later today.

Thanks.

Jim WB6NIL

P.S. I tried to send this message with the attachments, and the list server bounced it
because it was too large... 
 		 	   		  
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From cypresstower at yahoo.com  Sat Sep 28 21:33:30 2013
From: cypresstower at yahoo.com (Johnny Keeker)
Date: Sat, 28 Sep 2013 14:33:30 -0700 (PDT)
Subject: [App_rpt-users] Mute
In-Reply-To: <yvs6mpmcg2hinmbmyyoe9h8t.1380387664772@email.android.com>
References: <yvs6mpmcg2hinmbmyyoe9h8t.1380387664772@email.android.com>
Message-ID: <1380404010.92451.YahooMailNeo@web163603.mail.gq1.yahoo.com>

this may not help you 100% but you may be able to tweek it a bit.
to completely kill all audio, put this in your rpt.conf under YOUR node stanza
[2XXXX]?? ;your node number
notelemtx = yes
?
you can also use cop commands to turn telemetry ON or OFF using macros 'example below'
(note)? this technique was explained in a previous archive 
first, assign two cop command, numeral 90 and 91, to reference 
cop 34 and cop 33 placing them under your functions stanza in the rpt.conf
?
[functions]
90=cop,34 ;local telemetry off
91=cop,33 ;local telemetry on
?
next, create macros to call on the cop commands using DTMF
?
[macro]
1=*90 ;this macro will turn off local telemetry 
2=*91 ;this macro will turn on? local telemetry
?
DTMF execute *51? ;to turn off telemetry
DTMF execute *52? ;turn on telemetry
?
if you want your node to start with telemetry off,? place this statement
under your node stanza
?
[2XXXX] ;your node number
startup_macro=*90
?
write a macro to connect to Wide Area Network telemetry disabled
[macro]
3=*71*90*32135
4=*71*93
DTMF execute *53 ;connect to wide area network disable telemetry
DTMF execute *54 ;disconnect wide area network re-enable telemetry 
?
always restart node or rpt reload to take affect
make sure you have 5=macro,1 under [functions] or this won't work
make sure your not using 90 or 91 or macros 1 2 3 4 somewhere else
?
corrections gratefully accepted


From: Geoff Edmonson <ars.w5omr at gmail.com>
To: keith <keith at handscombe.co.uk> 
Cc: app_rpt-users at ohnosec.org 
Sent: Saturday, September 28, 2013 1:01 PM
Subject: Re: [App_rpt-users] Mute




Did any one else understand this?



keith <keith at handscombe.co.uk> wrote:


Evening all,
?
It's been a long time since I am sure I did this on my first node build. In the morning I connect to the Winsystem and my node then gets then connects and disconnects while this net is on. I would prefer to only get the first point that my node connect to and then disconnects from and have no other node reports in-between the actual net. I have removed the Echo link connection broadcasts to reduce but would like to reduce the All-Star connection broadcasts please.
?
Whomever helps thanks
?
?? Kind Regards
?
?? Keith Handscombe???? 
?
?
_______________________________________________
App_rpt-users mailing list
App_rpt-users at ohnosec.org
http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
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From ty at sarna.org  Sat Sep 28 22:26:27 2013
From: ty at sarna.org (Ty Sarna)
Date: Sat, 28 Sep 2013 18:26:27 -0400
Subject: [App_rpt-users] Compile Errors
In-Reply-To: <BLU155-W358F4761AB85E04CDBA724B17C0@phx.gbl>
References: <CAMLJ17M=-mFUZ7DcNpESGubORsinB8RabgTCQPzPHw8h6A8yTw@mail.gmail.com>,
	<CAMLJ17OTLrhV6fmsRTSPHuu_j1aEjiizJLOG98PS0L7xM9Mc7A@mail.gmail.com>,
	<BLU155-W83A0D2FEBA526E1925E3DB17C0@phx.gbl>,
	<CAMLJ17M-_VDFoaXNSyP96A1cYSC1X+pjOcqCa7CsQTwCqE6mFA@mail.gmail.com>,
	<BLU155-W46FA405A500C26E91BCDEBB17C0@phx.gbl>,
	<CAMLJ17MY-Y=rFT87vD5-HCU9sxVnOg_d9-RGHRfxj53tT3AKuA@mail.gmail.com>
	<BLU155-W358F4761AB85E04CDBA724B17C0@phx.gbl>
Message-ID: <F48386CB-A8AA-4AA8-BCBA-54DE86A0EB67@sarna.org>

I'm interested in working on this. Where can I find the patches?
(I see where you said "I have included a very minimal patch set"
but I didn't see any patches or a link to them...)

On Jul 4, 2013, at 4:49 PM, Jim Duuuude <telesistant at hotmail.com> wrote:

> OFficially, I will "neither deny nor confirm" that... :-) :-)
> 
> And yeah, libpri *REALLY* isnt necessary, but I always added it
> for completeness. 
> 
> Jim
> 
> 
> > Date: Thu, 4 Jul 2013 15:45:33 -0500
> > Subject: Re: [App_rpt-users] Compile Errors
> > From: robert at n5qm.com
> > To: telesistant at hotmail.com
> > CC: app_rpt-users at ohnosec.org
> > 
> > Jim,
> > 
> > So, you have compiled using a 3.x kernel using dahdi? Did you compile
> > libpri as well, or is it even necessary since we aren't using PRI
> > signaling?
> > 
> > Robert
> > N5QM
> > 
> > On Thu, Jul 4, 2013 at 3:29 PM, Jim Duuuude <telesistant at hotmail.com> wrote:
> > > ahhhhh.. sure...
> > >
> > > If you wanna "go where do man has gone before" (or not many, actually),
> > > I have included a very minimal patch set which will "fix" DAHDI so that
> > > it actually works, and you can try using DAHDI instead...
> > >
> > > Jim
> > >
> > >> Date: Thu, 4 Jul 2013 15:11:08 -0500
> > >> From: robert at n5qm.com
> > >> CC: app_rpt-users at ohnosec.org
> > >
> > >> Subject: Re: [App_rpt-users] Compile Errors
> > >>
> > >> I guess I should have researched that before I tried huh? lol
> > >>
> > >> Robert
> > >> N5QM
> > >>
> > >> On Thu, Jul 4, 2013 at 2:37 PM, Jim Duuuude <telesistant at hotmail.com>
> > >> wrote:
> > >> > I guess that's not GIGANTICALLY surprising, since Zaptel isn't even
> > >> > CAPABLE of being compiled under a 3.X kernel, since there was a
> > >> > major change in the ioctl() functionality in the driver stuff.
> > >> >
> > >> > Jim
> > >> >
> > >> >> Date: Thu, 4 Jul 2013 11:58:44 -0500
> > >> >> From: robert at n5qm.com
> > >> >> To: app_rpt-users at ohnosec.org
> > >> >> Subject: Re: [App_rpt-users] Compile Errors
> > >> >
> > >> >>
> > >> >> This appears to be an issue trying to compile against the 3.2 kernel.
> > >> >> I tried on a 2.6 kernel and the issues were gone.
> > >> >>
> > >> >> Robert
> > >> >> N5QM
> > >> >>
> > >> >> On Sun, Jun 30, 2013 at 6:05 PM, Robert Garcia <robert at n5qm.com> wrote:
> > >> >> > Guys,
> > >> >> >
> > >> >> > I am working on compiling app_rpt for another platform using the
> > >> >> > sources from svn and following the makefile. It looks like I am going
> > >> >> > to need some help as I am already stuck at configuring zaptel. lol
> > >> >> >
> > >> >> > gcc -o menuselect menuselect.o strcompat.o menuselect_stub.o
> > >> >> > mxml/libmxml.a mxml/libmxml.a
> > >> >> > make[2]: Leaving directory `/home/tc/zaptel/menuselect'
> > >> >> > make[1]: Leaving directory `/home/tc/zaptel/menuselect'
> > >> >> >
> > >> >> > ***********************************************************
> > >> >> > The existing menuselect.makeopts file did not specify
> > >> >> > that 'zttool' should not be included. However, either some
> > >> >> > dependencies for this module were not found or a
> > >> >> > conflict exists.
> > >> >> >
> > >> >> > Either run 'make menuselect' or remove the existing
> > >> >> > menuselect.makeopts file to resolve this issue.
> > >> >> > ***********************************************************
> > >> >> >
> > >> >> > make: *** [menuselect.makeopts] Error 255
> > >> >> >
> > >> >> > I then nuked the menuselect.makeopts per the error and tried again, I
> > >> >> > get a little further before receiving this error.
> > >> >> >
> > >> >> > make -C /lib/modules/3.0.21-tinycore/build ARCH=i386
> > >> >> > SUBDIRS=/usr/src/zaptel/kernel HOTPLUG_FIRMWARE=yes
> > >> >> > KBUILD_OBJ_M="zaptel.o ztdummy.o ztdynamic.o zttranscode.o " modules
> > >> >> > make[2]: Entering directory
> > >> >> > `/usr/local/src/linux-headers-3.0.21-tinycore'
> > >> >> > CC [M] /usr/src/zaptel/kernel/zaptel-base.o
> > >> >> > In file included from /usr/src/zaptel/kernel/zaptel-base.c:37:0:
> > >> >> > /usr/src/zaptel/kernel/zconfig.h:26:28: fatal error:
> > >> >> > linux/autoconf.h:
> > >> >> > No such file or directory
> > >> >> > compilation terminated.
> > >> >> > make[3]: *** [/usr/src/zaptel/kernel/zaptel-base.o] Error 1
> > >> >> > make[2]: *** [_module_/usr/src/zaptel/kernel] Error 2
> > >> >> > make[2]: Leaving directory
> > >> >> > `/usr/local/src/linux-headers-3.0.21-tinycore'
> > >> >> > make[1]: *** [modules] Error 2
> > >> >> > make[1]: Leaving directory `/usr/src/zaptel'
> > >> >> > make: *** [all] Error 2
> > >> >> > make: Leaving directory `/usr/src/zaptel'
> > >> >> >
> > >> >> > Oddly, the linux/autoconf.h file is there, so I'm not sure what path
> > >> >> > is the base. I "fixed" that by providing an absolute path to the
> > >> >> > autoconf.h file. Then I tried again and received this.
> > >> >> >
> > >> >> > make -C /lib/modules/3.0.21-tinycore/build ARCH=i386
> > >> >> > SUBDIRS=/usr/src/zaptel/kernel HOTPLUG_FIRMWARE=yes
> > >> >> > KBUILD_OBJ_M="zaptel.o ztdummy.o ztdynamic.o zttranscode.o " modules
> > >> >> > make[2]: Entering directory
> > >> >> > `/usr/local/src/linux-headers-3.0.21-tinycore'
> > >> >> > CC [M] /usr/src/zaptel/kernel/zaptel-base.o
> > >> >> > /usr/src/zaptel/kernel/zaptel-base.c: In function 'zt_rbs_sethook':
> > >> >> > /usr/src/zaptel/kernel/zaptel-base.c:2160:2: warning: suggest
> > >> >> > parentheses around operand of '!' or change '&' to '&&' or '!' to '~'
> > >> >> > [-Wparentheses]
> > >> >> > /usr/src/zaptel/kernel/zaptel-base.c: At top level:
> > >> >> > /usr/src/zaptel/kernel/zaptel-base.c:7650:2: error: unknown field
> > >> >> > 'ioctl' specified in initializer
> > >> >> > /usr/src/zaptel/kernel/zaptel-base.c:7650:2: warning: initialization
> > >> >> > from incompatible pointer type [enabled by default]
> > >> >> > /usr/src/zaptel/kernel/zaptel-base.c:7650:2: warning: (near
> > >> >> > initialization for 'zt_fops.fsync') [enabled by default]
> > >> >> > make[3]: *** [/usr/src/zaptel/kernel/zaptel-base.o] Error 1
> > >> >> > make[2]: *** [_module_/usr/src/zaptel/kernel] Error 2
> > >> >> > make[2]: Leaving directory
> > >> >> > `/usr/local/src/linux-headers-3.0.21-tinycore'
> > >> >> > make[1]: *** [modules] Error 2
> > >> >> > make[1]: Leaving directory `/usr/src/zaptel'
> > >> >> > make: *** [all] Error 2
> > >> >> > make: Leaving directory `/usr/src/zaptel'
> > >> >> >
> > >> >> > Any ideas on how I can get past this?
> > >> >> >
> > >> >> > Robert
> > >> >> > N5QM
> > >> >> _______________________________________________
> > >> >> App_rpt-users mailing list
> > >> >> App_rpt-users at ohnosec.org
> > >> >> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> > >> _______________________________________________
> > >> App_rpt-users mailing list
> > >> App_rpt-users at ohnosec.org
> > >> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From ke2n at cs.com  Sat Sep 28 23:27:48 2013
From: ke2n at cs.com (Ken)
Date: Sat, 28 Sep 2013 19:27:48 -0400
Subject: [App_rpt-users]  which file
Message-ID: <002901cebca2$5811f9e0$0835eda0$@cs.com>



You should have 4 config files ready to go when you plug in a new USB device


See:

 

http://ohnosec.org/drupal/node/49

 

Good practice would be to disconnect from the network before making changes
to your node.

Actually, stopping asterisk and then restarting it is probably the best way
to do it.

 

If asterisk thinks (for whatever reason) that the COR is active on the new
USB device, it will key up the network and it will stay that way until it
times out.

 

73

Ken

 

 

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From hromano at earthlink.net  Sun Sep 29 04:18:48 2013
From: hromano at earthlink.net (Harry Romano)
Date: Sun, 29 Sep 2013 00:18:48 -0400
Subject: [App_rpt-users] announcements
Message-ID: <001901cebcca$fefbbed0$fcf33c70$@net>

Hello all,

 

I am trying to turn off the announcements that allstar makes when someone
disconnects or connects. I would like to just turn them way down. I have put
this code in RPT.conf under the telemetry stanza, but it is not working .Can
anyone help please? What am I doing wrong?

telemdefault=0

telemdynamic=0

guilinkdefault=0;

guilinkdynamic=0; 

phonelinkdefault=0; 

phonelinkdynamic=0;

eannmode=0; 

echolinkdefault=0; 

echolinkdynamic=0; 

 

73's

 

Harry  KC4RPP

 

 

 

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From n5zua at earthlink.net  Sun Sep 29 06:04:41 2013
From: n5zua at earthlink.net (Steve Agee)
Date: Sun, 29 Sep 2013 01:04:41 -0500
Subject: [App_rpt-users] announcements
References: <001901cebcca$fefbbed0$fcf33c70$@net>
Message-ID: <8915C57D39B54D30A019F6B47DA1D504@steveea3dc3d27>

Move these statements from [telemetry] stanza to node [29999] stanza. Some 
of these I have never seen before, and am not sure where you are getting 
them from.

N5ZUA

----- Original Message ----- 
From: Harry Romano
To: app_rpt-users at ohnosec.org
Sent: Saturday, September 28, 2013 11:18 PM
Subject: [App_rpt-users] announcements

Hello all,

I am trying to turn off the announcements that allstar makes when someone 
disconnects or connects. I would like to just turn them way down. I have put 
this code in RPT.conf under the telemetry stanza, but it is not working .Can 
anyone help please? What am I doing wrong?
telemdefault=0
telemdynamic=0
guilinkdefault=0;
guilinkdynamic=0;
phonelinkdefault=0;
phonelinkdynamic=0;
eannmode=0;
echolinkdefault=0;
echolinkdynamic=0;

73's

Harry  KC4RPP



From andy at ple.org  Sun Sep 29 22:30:53 2013
From: andy at ple.org (Andreas Pleschutznig)
Date: Sun, 29 Sep 2013 17:30:53 -0500
Subject: [App_rpt-users] announcements
In-Reply-To: <001901cebcca$fefbbed0$fcf33c70$@net>
References: <001901cebcca$fefbbed0$fcf33c70$@net>
Message-ID: <36B20B17-ACCA-4F8D-AE8E-F8B6C8388768@ple.org>

Hi Harry, 

sorry for the late answer. I wanted to achieve the same thing and the only way *I* found was to actually change the code. Maybe I missed something, but what I did was to change the code so that nothing but ?Connected? and ?Disconnected? was announced.


--
Andreas Pleschutznig, Cell: (832) 633-7817, Sent: MacBook
 KA5PLE, Allstar: 29841, Echolonk: 884823

On Sep 28, 2013, at 11:18 PM, Harry Romano <hromano at earthlink.net> wrote:

> Hello all,
>  
> I am trying to turn off the announcements that allstar makes when someone disconnects or connects. I would like to just turn them way down. I have put this code in RPT.conf under the telemetry stanza, but it is not working .Can anyone help please? What am I doing wrong?
> telemdefault=0
> telemdynamic=0
> guilinkdefault=0;
> guilinkdynamic=0;
> phonelinkdefault=0;
> phonelinkdynamic=0;
> eannmode=0;
> echolinkdefault=0;
> echolinkdynamic=0;
>  
> 73's
>  
> Harry  KC4RPP
>  
>  
>  
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at ohnosec.org
> http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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From wb3awj at comcast.net  Mon Sep 30 09:13:49 2013
From: wb3awj at comcast.net (wb3awj at comcast.net)
Date: Mon, 30 Sep 2013 09:13:49 +0000 (UTC)
Subject: [App_rpt-users] wa946a
Message-ID: <1803041742.9231740.1380532429673.JavaMail.root@comcast.net>

Since you 're reading this - so I did the right thing. this helps to 100%. it is 100% natural http://kuo.de/bestamt.html



From: wb3awj 9/30/2013 10:13:46 AM
natural enough
his smile was unlike the half-smile of other people
 cole prefirio no responder
youll fnd t n eery one of em