From k1lnx at k1lnx.net  Wed Feb  3 00:42:10 2010
From: k1lnx at k1lnx.net (Stephen - K1LNX)
Date: Tue, 2 Feb 2010 19:42:10 -0500
Subject: [App_rpt-users] Reverse Autopatch radio PTT
Message-ID: <8390870f1002021642l3631451el6533ca4265597239@mail.gmail.com>

I finally got my main PBX peered to my app_rpt box (long story) and I set up
normal phone control mode to have full functionality. With simple phone
control, I can key/unkey the transmitter with * and # respectively.

How do I accomplish that in normal phone control mode? Also, if I connect to
an echolink or allstar node from my phone, how does audio (if it can be) get
sent to the network?

For example, say I connect to my node via phone and I want to connect to
Allstar node 2000. Can I send audio from my phone? I am finding it very
useful to monitor things, but the ability to transmit audio to the network
would be rather neat.....

Is this possible or is this beyond the scope of design?

tnx and 73
Stephen
K1LNX

-- 
**********************************
Stephen Brown - ARS K1LNX
Johnson City, TN EM86
http://www.k1lnx.net
google voice: 423-665-9367
**********************************
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From hkwilliamson at gmail.com  Wed Feb  3 02:22:27 2010
From: hkwilliamson at gmail.com (Keith Williamson)
Date: Tue, 2 Feb 2010 19:22:27 -0700
Subject: [App_rpt-users] Reverse Autopatch radio PTT
In-Reply-To: <8390870f1002021642l3631451el6533ca4265597239@mail.gmail.com>
References: <8390870f1002021642l3631451el6533ca4265597239@mail.gmail.com>
Message-ID: <d15872231002021822r245b9e90m21cb55eeb8a953c1@mail.gmail.com>

Hi Stephen,

I'm using normal mode of reverse autopatch and by default, you should
be able to key up with *98 and un-key with #. Of course you can change
those to something else and the best way to do that so you don't
affect regular users used to the existing mappings is to edit rpt.conf
and dup the entire [funtions] stanza creating a second one called
[p_functions] and then change the line earlier in the file that maps
phone_funtions from funtions to p_functions. If you look at the file
you'll see what I mean.

Cheers,

Keith

KF7DRV
Node 2541

On 2/2/10, Stephen - K1LNX <k1lnx at k1lnx.net> wrote:
> I finally got my main PBX peered to my app_rpt box (long story) and I set up
> normal phone control mode to have full functionality. With simple phone
> control, I can key/unkey the transmitter with * and # respectively.
>
> How do I accomplish that in normal phone control mode? Also, if I connect to
> an echolink or allstar node from my phone, how does audio (if it can be) get
> sent to the network?
>
> For example, say I connect to my node via phone and I want to connect to
> Allstar node 2000. Can I send audio from my phone? I am finding it very
> useful to monitor things, but the ability to transmit audio to the network
> would be rather neat.....
>
> Is this possible or is this beyond the scope of design?
>
> tnx and 73
> Stephen
> K1LNX
>
> --
> **********************************
> Stephen Brown - ARS K1LNX
> Johnson City, TN EM86
> http://www.k1lnx.net
> google voice: 423-665-9367
> **********************************
>

-- 
Sent from my mobile device


From hkwilliamson at gmail.com  Wed Feb  3 02:52:12 2010
From: hkwilliamson at gmail.com (Keith Williamson)
Date: Tue, 2 Feb 2010 19:52:12 -0700
Subject: [App_rpt-users] Example of how to setup a sip phone for reverse
	autopatch
Message-ID: <d15872231002021852i282f18d8wffb637d538247d2f@mail.gmail.com>

Thought I'd share this thread with the list since it might be helpful to
someone trying to setup a local sip phone for reverse autopatch.

Bob and I went off the list when we first got started on this because I was
afraid the thread could get quite lengthy if we ran into the (almost)
inevitable problems. However, everything worked the first time through a
combination of luck on my part explaining things (blue moon) and Bob's
excellent ability to read, interpret, and follow my drivel.

Bob, if there were things I explained incorrectly below that you figured
your way around, please chime in so that the next unwary ham who tries this
will be forewarned. Also note that some of this is specific to Bob's SIP
phone but in looking through the manual for the phone, I found it to be
pretty typical of other SIP phones I've worked with. Finally, apologies for
the reverse ordering of the thread versus the way we normally (try) to order
thread (newest reply at bottom).

73's

Keith

KF7DRV
Allstar Node 2541

---------- Forwarded message ----------
From: Bob Brown - W?NQX <bbrown at byrg.net>
Date: Sun, Jan 31, 2010 at 12:03 PM
Subject: SIP IP pHONE config - was: Connect notifications via AGI or System
scripts
To: Keith Williamson <hkwilliamson at gmail.com>


hi keith

got the first part done
~~~~~~~~~~~~~~~~~~~~~~
sip show peers

Name/username              Host            Dyn Nat ACL Port
Status
9009/9009                  10.0.0.40        D          5060
Unmonitored
1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0
offline]
 ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

whats next?

--
Thanks in Advance

  Bob Brown, W?NQX

  Kansas City Metro Area

  http://sm0kenet.net

  http://byrg.net

  http://kcdstar.byrg.net

Quis custodiet ipsos custodes?



On Sun, Jan 31, 2010 at 12:09, Keith Williamson <hkwilliamson at gmail.com>wrote:

> OK great. Next step is to get the phone registered with the Allstar node.
> I'm going to assume the phone is connected to the internal network the node
> is on and for the purposes of this explanation I'll say the internal IP of
> your Allstar node is 192.168.0.1 (and that that IP is either static or gets
> a static lease via DHCP).  Obviously, substitute your node's actual internal
> IP.
>
> In the web interface for the phone, enter the Allstar node internal IP in
> ALL of the following fields of Line 1 settings for Profile 1:
>
> SIP Proxy Server
> Outbound Proxy Server
> Registrar Server
> Outbound Registrar Server
>
> Leave the SIP Proxy Server Port and Registrar Proxy Server Port set at the
> default of 5060.
>
> Also in the same Line 1 Profile 1 menu, you need to set the phone's
> extension info. You can make this whatever you want but for the purposes of
> this explanation we'll call it Bobs208. Set Bobs208 in the following fields:
>
> Phone Number
> User Name
> Authorized ID
>
> Finally, you need to set a password for authentication. We'll use
> "secret123". Set the following field to secret123:
>
> Authorized Password
>
> This should be enough on the phone side of things so now down at the bottom
> of the menu, do Save Settings followed by Logout.
>
> Now we move over to the Allstar node. Open /etc/asterisk/sip.conf with your
> favorite editor (I'm a vi guy myself) and add a stanza for your phone (below
> the end of the [general] stanza) that would be something like this:
>
> [Bobs208]
> type=friend
> secret=secret123
> context=radio-control
> host=dynamic
> disallow=all
> allow=ulaw
> transfer=no
>
> Now save the changes to sip.conf and exit the editor. Next go into the CLI
> (e.g. asterisk -vr) and type "sip reload" to force asterisk to reread the
> sip.conf file. At this point, asterisk should be ready register your phone.
>
> Probably the most dependable way to get the phone to activate the changes
> you made to the phone's SIP parameters and to force the phone to attempt to
> register with Allstar/Asterisk is to just power-cycle the phone.
>
> I haven't read enough about this phone to know how the phone indicates it's
> registration state after it finishes booting but you can watch the Allstar
> CLI and you should be able to see the registration occuring. You can also
> type the following CLI command to check if you phone is registered:
>
> CLI> sip show peers
>
> Let me know how all this goes and then we can proceed to the final step
> once the phone is registered.
>
> 73's
>
> Keith
>
> KF7DRV
>
>
>
>
>
> On Sat, Jan 30, 2010 at 11:57 PM, Bob Brown - W?NQX <bbrown at byrg.net>wrote:
>
>> ya mine is a virgin fone and the default is 1234
>>
>> i got this from a ip phone vendor/wholsaler it was a freebie he says it
>> was sip compatable
>>
>> i have managed to log into it from the web interface and from the admin
>> menu in the front panle.
>>
>>
>>
>> --
>> Thanks in Advance
>>
>>   Bob Brown, W?NQX
>>
>>   Kansas City Metro Area
>>
>>   http://sm0kenet.net
>>
>>   http://byrg.net
>>
>>   http://kcdstar.byrg.net
>>
>> Quis custodiet ipsos custodes?
>>
>>
>>
>>   On Sat, Jan 30, 2010 at 22:45, Keith Williamson <hkwilliamson at gmail.com
>> > wrote:
>>
>>> Hi Bob,
>>>
>>> I found the manual for your T207/T208 phone..hopefully you have that too.
>>> Getting the phone registered with Asterisk requires that the phone be an "S"
>>> model (T207S or T208S) which means it has SIP firmware. The T207M and T208M
>>> models run MGCP which isn't compatible with Asterisk. The other requirement
>>> is that you need to know the admin password. If it's never been changed it
>>> should be the default "1234". You can configure the phone settings either
>>> directly on the phone or via your web browser (although you need to at least
>>> configure the TCP/IP parameters using the phone menus before you can switch
>>> to using the more convenient web interface).
>>>
>>> Perhaps you have already done this part and have the phone connected via
>>> Ethernet to the network the node is on?
>>>
>>> Let me know and we can go from there.
>>>
>>> 73s,
>>>
>>> Keith
>>>
>>>
>>> On Sat, Jan 30, 2010 at 10:02 AM, Bob Brown - W?NQX <bbrown at byrg.net>wrote:
>>>
>>>> my ipphone: TSIPP2008BG-R1
>>>>
>>>> sure any help would be great!
>>>>
>>>>
>>>>
>>>> --
>>>> Thanks in Advance
>>>>
>>>>   Bob Brown, W?NQX
>>>>
>>>>   Kansas City Metro Area
>>>>
>>>>   http://sm0kenet.net
>>>>
>>>>   http://byrg.net
>>>>
>>>>   http://kcdstar.byrg.net
>>>>
>>>> Quis custodiet ipsos custodes?
>>>>
>>>>
>>>>
>>>>   On Fri, Jan 29, 2010 at 23:01, Keith Williamson <
>>>> hkwilliamson at gmail.com> wrote:
>>>>
>>>>> Hi Bob,
>>>>>
>>>>> No problem. What make/model of ipphone is it? The first thing required
>>>>> is a stanza in sip.conf defining the extension, user, password, etc
>>>>> for that particular phone. The required elements vary from phone to
>>>>> phone so let me know what kind you have. After creating the entry in
>>>>> sip.conf, you just do a "sip reload" at the CLI and check to see if
>>>>> the phone gets registered OK. Once you have that, you just need to add
>>>>> a stanza to extensions.conf to enable the reverse autopatch.
>>>>>
>>>>> There's a pretty complete description of what's required in the ACID
>>>>> Admin Manual.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Keith
>>>>>
>>>>>
>>>>> On 1/29/10, Bob Brown - W?NQX <bbrown at byrg.net> wrote:
>>>>> > hi keith
>>>>> >
>>>>> > i would be interrested in your config files to set up the ipphone
>>>>> > on your node
>>>>> >
>>>>> > i have a 4 line ip desk phone i would like to set up to do what you
>>>>> describe
>>>>> >
>>>>> > i am not very fluent in asterisk set up for ip fones.
>>>>> >
>>>>> > would you be willing to share?
>>>>> >
>>>>> >
>>>>> > --
>>>>> > Thanks in Advance
>>>>> >
>>>>> >   Bob Brown, W?NQX
>>>>> >
>>>>> >   Kansas City Metro Area
>>>>> >
>>>>> >   http://sm0kenet.net
>>>>> >
>>>>> >   http://byrg.net
>>>>> >
>>>>> >   http://kcdstar.byrg.net
>>>>> >
>>>>> > Quis custodiet ipsos custodes?
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Fri, Jan 29, 2010 at 22:04, Keith Williamson
>>>>> > <hkwilliamson at gmail.com>wrote:
>>>>> >
>>>>> >> Hi,
>>>>> >>
>>>>> >> Recently, I got reverse autopatch configured on my simplex node
>>>>> (2541) and
>>>>> >> everything works great. In the shack, I have a Polycom IP501 speaker
>>>>> phone
>>>>> >> that I use to monitor or connect into QSO's on the node. Out of the
>>>>> shack,
>>>>> >> I
>>>>> >> can call into the node using my Blackberry and do the same. So the
>>>>> next
>>>>> >> challenge is to get quick notifications of users connecting into the
>>>>> node
>>>>> >> either via radio or Internet (Echolink and Allstar..no IRLP yet). I
>>>>> >> created
>>>>> >> a pair of Twitter accounts; one for the node and another for me
>>>>> >> personally.
>>>>> >> I added my node's Twitter account to the ones I follow with my
>>>>> personal
>>>>> >> account and created an AGI scripts that formats a curl command which
>>>>> posts
>>>>> >> some of the context variables passed in when Asterisk invokes the
>>>>> script
>>>>> >> (context name, extension, callerid, etc). I then added the AGI call
>>>>> into
>>>>> >> various extension stanzas in extensions.conf to test. If, for
>>>>> instance, I
>>>>> >> do
>>>>> >> a reverse autopatch and connect to the reverse autopatch
>>>>> "extension", I
>>>>> >> almost immediately get an SMS tweet on the Blackberry.  Great.
>>>>> However, it
>>>>> >> seems that connections via the radio are not processed in any way in
>>>>> >> extensions.conf (thought [default] stanza would apply but apparently
>>>>> only
>>>>> >> applies to autopatch), and inbound connections from either Allstar
>>>>> or
>>>>> >> Echolink, while processed through [radio-secure], hang on the AGI
>>>>> call and
>>>>> >> don't proceed to the following call to rpt. I'm assuming this is
>>>>> because
>>>>> >> AGI
>>>>> >> calls generally follow an "Answer" and are only valid in the
>>>>> connected
>>>>> >> state. I tried "deadAGI" since it doesn't seems to be dependent on
>>>>> being
>>>>> >> in
>>>>> >> the answered state but it hung the same way. It's possible I've got
>>>>> a
>>>>> >> problem in the AGI script which is causing the hang but it certainly
>>>>> >> doesn't
>>>>> >> occur when the AGI call follows "Answer".
>>>>> >>
>>>>> >> So I'm looking for help to understand other possible ways to invoke
>>>>> a
>>>>> >> script that calls curl to post the tweet when a radio user makes an
>>>>> >> outbound
>>>>> >> connection to Allstar or Echolink and when an Echolink or Allstar
>>>>> user
>>>>> >> connects in to the radio.
>>>>> >>
>>>>> >> Ideas?
>>>>> >>
>>>>> >> Thanks and 73's
>>>>> >>
>>>>> >> Keith
>>>>> >>
>>>>> >> KF7DRV
>>>>> >>
>>>>> >> _______________________________________________
>>>>> >> App_rpt-users mailing list
>>>>> >> App_rpt-users at qrvc.com
>>>>> >> http://qrvc.com/mailman/listinfo/app_rpt-users
>>>>> >>
>>>>> >>
>>>>> >
>>>>>
>>>>> --
>>>>> Sent from my mobile device
>>>>>
>>>>
>>>>
>>>
>>
>
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From k1lnx at k1lnx.net  Wed Feb  3 04:27:48 2010
From: k1lnx at k1lnx.net (Stephen - K1LNX)
Date: Tue, 2 Feb 2010 23:27:48 -0500
Subject: [App_rpt-users] Example of how to setup a sip phone for reverse
	autopatch
In-Reply-To: <d15872231002021852i282f18d8wffb637d538247d2f@mail.gmail.com>
References: <d15872231002021852i282f18d8wffb637d538247d2f@mail.gmail.com>
Message-ID: <8390870f1002022027m1aa78b2dwef1ed762e385fcdd@mail.gmail.com>

Thanks Keith for sharing that with us, it will certainly help :)

I'll also add my own experiences to this for others who may be trying to
accomplish what I have as well. I run a seperate Asterisk based PBX for my
home phone system which I have 2 Cisco IP phone's peered off of.

My ultimate goal was to be able to use my existing phone system to get to
the radio and out to the network. I use FreePBX on the PBX box, and just set
up a SIP trunk to point to the app_rpt box. In the outgoing settings area,
you can name the trunk anything you'd like (I named mine app_rpt, original
huh?) and in the peer details box I have this:

host=192.168.1.1
username=k1lnx
secret=password
type=peer

Where host is the IP of the app_rpt machine, username is the username you
will define in sip.conf stanza, secret is the password, and type=peer
defines that we are only sending calls to this machine.

You'll also need to put registration syntax in the Register String box in
the form of username:password at 192.168.1.1/context which was defined in the
previous section. The "context" portion of the string is what your context
name will be on the app_rpt box. For example, mine is k1lnx. You could also
do IP based authentication, which would eliminate the need to register.

Next, add an outbound route with whatever dial pattern you want to match (I
just used 2335, that's my node number, to keep it simple) and use the newly
created app_rpt SIP trunk. I named my route RADIO. This will now send any
calls dialed with "2335" over the app_rpt trunk!

So on the app_rpt box in /etc/asterisk/sip.conf put an entry something
similar to this:

[k1lnx]
type=friend
username=k1lnx
password=password
context=radio-control
host=dynamic
disallow=all
allow=ulaw

And in extensions.conf edit (or add if not existing) the [radio-control]
stanza:
[radio-control]
;exten => 2335,1,Answer
exten => 2335,1,Playback(rpt/node)
exten => 2335,n,Playback(digits/2)
exten => 2335,n,Playback(digits/3)
exten => 2335,n,Playback(digits/3)
exten => 2335,n,Playback(digits/5)
exten => 2335,n,Rpt,2335|P

Notice I have the first priority commented out? This was due to an issue I
was having, the console was spewing error messages about not being able to
use the Answer application. No idea why, could be due to the fact that I
compiled from scratch or screwed something else up somewhere that I haven't
found yet. YMMV ;)

I also had a hard time getting calls from my PBX to hit this context
properly, it took a lot of wrangling and reloading for it to finally work
properly. Whenever I would dial "2335" from my desk phone it would get
rejected on the basis of not being able to find the extension, I am still
largely confused as to why it didn't simply work however. On the stock
extensions.conf that comes with ACID as well as what is in the svn version
at the top of the file there are two parameters in the general section:

[general]

static=yes       ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.

I had to play with these via commenting/uncommenting them to get things to
work, although I don't believe this was the source of my problem originally.
Perhaps Steve or Jim can elaborate on this.

After you issue a dialplan reload you can check for the presence of your
newly created extension by issuing "dialplan show extension at context" from
the CLI, where extension is the extension number, and context is the context
you created (radio-control in my example), and it will return the routing
for that extension:

bubastis*CLI> dialplan show 2335 at radio-control
[ Context 'radio-control' created by 'pbx_config' ]
  '2335' =>         1. Playback(rpt/node)
[pbx_config]
                    2. Playback(digits/2)
[pbx_config]
                    3. Playback(digits/3)
[pbx_config]
                    4. Playback(digits/3)
[pbx_config]
                    5. Playback(digits/5)
[pbx_config]
                    6. Rpt(2335|P )
[pbx_config]

-= 1 extension (6 priorities) in 1 context. =-

There are other methods to accomplish what I have above, and of course
security concerns. This works extremely well and now that I have it working
I am going to continue to tweak things and make it as robust as I possibly
can. My next challenge is to get inbound and outbound autopatch calls from
the app_rpt box to my provider.

Love this project, I've been playing with it for just over a year now and I
am still learning stuff. So much fun...

tnx and 73
Stephen
K1LNX


On Tue, Feb 2, 2010 at 9:52 PM, Keith Williamson <hkwilliamson at gmail.com>wrote:

> Thought I'd share this thread with the list since it might be helpful to
> someone trying to setup a local sip phone for reverse autopatch.
>
> Bob and I went off the list when we first got started on this because I was
> afraid the thread could get quite lengthy if we ran into the (almost)
> inevitable problems. However, everything worked the first time through a
> combination of luck on my part explaining things (blue moon) and Bob's
> excellent ability to read, interpret, and follow my drivel.
>
> Bob, if there were things I explained incorrectly below that you figured
> your way around, please chime in so that the next unwary ham who tries this
> will be forewarned. Also note that some of this is specific to Bob's SIP
> phone but in looking through the manual for the phone, I found it to be
> pretty typical of other SIP phones I've worked with. Finally, apologies for
> the reverse ordering of the thread versus the way we normally (try) to order
> thread (newest reply at bottom).
>
> 73's
>
> Keith
>
> KF7DRV
> Allstar Node 2541
>
> ---------- Forwarded message ----------
> From: Bob Brown - W?NQX <bbrown at byrg.net>
> Date: Sun, Jan 31, 2010 at 12:03 PM
> Subject: SIP IP pHONE config - was: Connect notifications via AGI or System
> scripts
> To: Keith Williamson <hkwilliamson at gmail.com>
>
>
> hi keith
>
> got the first part done
> ~~~~~~~~~~~~~~~~~~~~~~
> sip show peers
>
> Name/username              Host            Dyn Nat ACL Port
> Status
> 9009/9009                  10.0.0.40        D          5060
> Unmonitored
> 1 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 0
> offline]
>  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
>
> whats next?
>
> --
> Thanks in Advance
>
>   Bob Brown, W?NQX
>
>   Kansas City Metro Area
>
>   http://sm0kenet.net
>
>   http://byrg.net
>
>   http://kcdstar.byrg.net
>
> Quis custodiet ipsos custodes?
>
>
>
> On Sun, Jan 31, 2010 at 12:09, Keith Williamson <hkwilliamson at gmail.com>wrote:
>
>> OK great. Next step is to get the phone registered with the Allstar node.
>> I'm going to assume the phone is connected to the internal network the node
>> is on and for the purposes of this explanation I'll say the internal IP of
>> your Allstar node is 192.168.0.1 (and that that IP is either static or gets
>> a static lease via DHCP).  Obviously, substitute your node's actual internal
>> IP.
>>
>> In the web interface for the phone, enter the Allstar node internal IP in
>> ALL of the following fields of Line 1 settings for Profile 1:
>>
>> SIP Proxy Server
>> Outbound Proxy Server
>> Registrar Server
>> Outbound Registrar Server
>>
>> Leave the SIP Proxy Server Port and Registrar Proxy Server Port set at the
>> default of 5060.
>>
>> Also in the same Line 1 Profile 1 menu, you need to set the phone's
>> extension info. You can make this whatever you want but for the purposes of
>> this explanation we'll call it Bobs208. Set Bobs208 in the following fields:
>>
>> Phone Number
>> User Name
>> Authorized ID
>>
>> Finally, you need to set a password for authentication. We'll use
>> "secret123". Set the following field to secret123:
>>
>> Authorized Password
>>
>> This should be enough on the phone side of things so now down at the
>> bottom of the menu, do Save Settings followed by Logout.
>>
>> Now we move over to the Allstar node. Open /etc/asterisk/sip.conf with
>> your favorite editor (I'm a vi guy myself) and add a stanza for your phone
>> (below the end of the [general] stanza) that would be something like this:
>>
>> [Bobs208]
>> type=friend
>> secret=secret123
>> context=radio-control
>> host=dynamic
>> disallow=all
>> allow=ulaw
>> transfer=no
>>
>> Now save the changes to sip.conf and exit the editor. Next go into the CLI
>> (e.g. asterisk -vr) and type "sip reload" to force asterisk to reread the
>> sip.conf file. At this point, asterisk should be ready register your phone.
>>
>> Probably the most dependable way to get the phone to activate the changes
>> you made to the phone's SIP parameters and to force the phone to attempt to
>> register with Allstar/Asterisk is to just power-cycle the phone.
>>
>> I haven't read enough about this phone to know how the phone indicates
>> it's registration state after it finishes booting but you can watch the
>> Allstar CLI and you should be able to see the registration occuring. You can
>> also type the following CLI command to check if you phone is registered:
>>
>> CLI> sip show peers
>>
>> Let me know how all this goes and then we can proceed to the final step
>> once the phone is registered.
>>
>> 73's
>>
>> Keith
>>
>> KF7DRV
>>
>>
>>
>>
>>
>> On Sat, Jan 30, 2010 at 11:57 PM, Bob Brown - W?NQX <bbrown at byrg.net>wrote:
>>
>>> ya mine is a virgin fone and the default is 1234
>>>
>>> i got this from a ip phone vendor/wholsaler it was a freebie he says it
>>> was sip compatable
>>>
>>> i have managed to log into it from the web interface and from the admin
>>> menu in the front panle.
>>>
>>>
>>>
>>> --
>>> Thanks in Advance
>>>
>>>   Bob Brown, W?NQX
>>>
>>>   Kansas City Metro Area
>>>
>>>   http://sm0kenet.net
>>>
>>>   http://byrg.net
>>>
>>>   http://kcdstar.byrg.net
>>>
>>> Quis custodiet ipsos custodes?
>>>
>>>
>>>
>>>   On Sat, Jan 30, 2010 at 22:45, Keith Williamson <
>>> hkwilliamson at gmail.com> wrote:
>>>
>>>> Hi Bob,
>>>>
>>>> I found the manual for your T207/T208 phone..hopefully you have that
>>>> too. Getting the phone registered with Asterisk requires that the phone be
>>>> an "S" model (T207S or T208S) which means it has SIP firmware. The T207M and
>>>> T208M models run MGCP which isn't compatible with Asterisk. The other
>>>> requirement is that you need to know the admin password. If it's never been
>>>> changed it should be the default "1234". You can configure the phone
>>>> settings either directly on the phone or via your web browser (although you
>>>> need to at least configure the TCP/IP parameters using the phone menus
>>>> before you can switch to using the more convenient web interface).
>>>>
>>>> Perhaps you have already done this part and have the phone connected via
>>>> Ethernet to the network the node is on?
>>>>
>>>> Let me know and we can go from there.
>>>>
>>>> 73s,
>>>>
>>>> Keith
>>>>
>>>>
>>>> On Sat, Jan 30, 2010 at 10:02 AM, Bob Brown - W?NQX <bbrown at byrg.net>wrote:
>>>>
>>>>> my ipphone: TSIPP2008BG-R1
>>>>>
>>>>> sure any help would be great!
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Thanks in Advance
>>>>>
>>>>>   Bob Brown, W?NQX
>>>>>
>>>>>   Kansas City Metro Area
>>>>>
>>>>>   http://sm0kenet.net
>>>>>
>>>>>   http://byrg.net
>>>>>
>>>>>   http://kcdstar.byrg.net
>>>>>
>>>>> Quis custodiet ipsos custodes?
>>>>>
>>>>>
>>>>>
>>>>>   On Fri, Jan 29, 2010 at 23:01, Keith Williamson <
>>>>> hkwilliamson at gmail.com> wrote:
>>>>>
>>>>>> Hi Bob,
>>>>>>
>>>>>> No problem. What make/model of ipphone is it? The first thing required
>>>>>> is a stanza in sip.conf defining the extension, user, password, etc
>>>>>> for that particular phone. The required elements vary from phone to
>>>>>> phone so let me know what kind you have. After creating the entry in
>>>>>> sip.conf, you just do a "sip reload" at the CLI and check to see if
>>>>>> the phone gets registered OK. Once you have that, you just need to add
>>>>>> a stanza to extensions.conf to enable the reverse autopatch.
>>>>>>
>>>>>> There's a pretty complete description of what's required in the ACID
>>>>>> Admin Manual.
>>>>>>
>>>>>> Cheers,
>>>>>>
>>>>>> Keith
>>>>>>
>>>>>>
>>>>>> On 1/29/10, Bob Brown - W?NQX <bbrown at byrg.net> wrote:
>>>>>> > hi keith
>>>>>> >
>>>>>> > i would be interrested in your config files to set up the ipphone
>>>>>> > on your node
>>>>>> >
>>>>>> > i have a 4 line ip desk phone i would like to set up to do what you
>>>>>> describe
>>>>>> >
>>>>>> > i am not very fluent in asterisk set up for ip fones.
>>>>>> >
>>>>>> > would you be willing to share?
>>>>>> >
>>>>>> >
>>>>>> > --
>>>>>> > Thanks in Advance
>>>>>> >
>>>>>> >   Bob Brown, W?NQX
>>>>>> >
>>>>>> >   Kansas City Metro Area
>>>>>> >
>>>>>> >   http://sm0kenet.net
>>>>>> >
>>>>>> >   http://byrg.net
>>>>>> >
>>>>>> >   http://kcdstar.byrg.net
>>>>>> >
>>>>>> > Quis custodiet ipsos custodes?
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> > On Fri, Jan 29, 2010 at 22:04, Keith Williamson
>>>>>> > <hkwilliamson at gmail.com>wrote:
>>>>>> >
>>>>>> >> Hi,
>>>>>> >>
>>>>>> >> Recently, I got reverse autopatch configured on my simplex node
>>>>>> (2541) and
>>>>>> >> everything works great. In the shack, I have a Polycom IP501
>>>>>> speaker phone
>>>>>> >> that I use to monitor or connect into QSO's on the node. Out of the
>>>>>> shack,
>>>>>> >> I
>>>>>> >> can call into the node using my Blackberry and do the same. So the
>>>>>> next
>>>>>> >> challenge is to get quick notifications of users connecting into
>>>>>> the node
>>>>>> >> either via radio or Internet (Echolink and Allstar..no IRLP yet). I
>>>>>> >> created
>>>>>> >> a pair of Twitter accounts; one for the node and another for me
>>>>>> >> personally.
>>>>>> >> I added my node's Twitter account to the ones I follow with my
>>>>>> personal
>>>>>> >> account and created an AGI scripts that formats a curl command
>>>>>> which posts
>>>>>> >> some of the context variables passed in when Asterisk invokes the
>>>>>> script
>>>>>> >> (context name, extension, callerid, etc). I then added the AGI call
>>>>>> into
>>>>>> >> various extension stanzas in extensions.conf to test. If, for
>>>>>> instance, I
>>>>>> >> do
>>>>>> >> a reverse autopatch and connect to the reverse autopatch
>>>>>> "extension", I
>>>>>> >> almost immediately get an SMS tweet on the Blackberry.  Great.
>>>>>> However, it
>>>>>> >> seems that connections via the radio are not processed in any way
>>>>>> in
>>>>>> >> extensions.conf (thought [default] stanza would apply but
>>>>>> apparently only
>>>>>> >> applies to autopatch), and inbound connections from either Allstar
>>>>>> or
>>>>>> >> Echolink, while processed through [radio-secure], hang on the AGI
>>>>>> call and
>>>>>> >> don't proceed to the following call to rpt. I'm assuming this is
>>>>>> because
>>>>>> >> AGI
>>>>>> >> calls generally follow an "Answer" and are only valid in the
>>>>>> connected
>>>>>> >> state. I tried "deadAGI" since it doesn't seems to be dependent on
>>>>>> being
>>>>>> >> in
>>>>>> >> the answered state but it hung the same way. It's possible I've got
>>>>>> a
>>>>>> >> problem in the AGI script which is causing the hang but it
>>>>>> certainly
>>>>>> >> doesn't
>>>>>> >> occur when the AGI call follows "Answer".
>>>>>> >>
>>>>>> >> So I'm looking for help to understand other possible ways to invoke
>>>>>> a
>>>>>> >> script that calls curl to post the tweet when a radio user makes an
>>>>>> >> outbound
>>>>>> >> connection to Allstar or Echolink and when an Echolink or Allstar
>>>>>> user
>>>>>> >> connects in to the radio.
>>>>>> >>
>>>>>> >> Ideas?
>>>>>> >>
>>>>>> >> Thanks and 73's
>>>>>> >>
>>>>>> >> Keith
>>>>>> >>
>>>>>> >> KF7DRV
>>>>>> >>
>>>>>> >> _______________________________________________
>>>>>> >> App_rpt-users mailing list
>>>>>> >> App_rpt-users at qrvc.com
>>>>>> >> http://qrvc.com/mailman/listinfo/app_rpt-users
>>>>>> >>
>>>>>> >>
>>>>>> >
>>>>>>
>>>>>> --
>>>>>> Sent from my mobile device
>>>>>>
>>>>>
>>>>>
>>>>
>>>
>>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
>


-- 
**********************************
Stephen Brown - ARS K1LNX
Johnson City, TN EM86
http://www.k1lnx.net
google voice: 423-665-9367
**********************************
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From monty at ke7jvx.com  Wed Feb  3 07:28:35 2010
From: monty at ke7jvx.com (Monty Dana)
Date: Wed, 3 Feb 2010 00:28:35 -0700
Subject: [App_rpt-users] Private Network
In-Reply-To: <14bc5ca01002011240n3c613a3dyda0d0005b81f6c66@mail.gmail.com>
References: <14bc5ca01002011240n3c613a3dyda0d0005b81f6c66@mail.gmail.com>
Message-ID: <14bc5ca01002022328v78d59f88gfc218aeb467f2cf8@mail.gmail.com>

Hello Everyone!

I wondering how one might set up a private network of connected nodes?
Right now I have two nodes setup here at my home, and would like to
experiment with having them both tied together on my private home network,
and still have one node able to access allstar and echolink, is that
possible?

What the main reason for this question is that many of you know, broadband
is hard to come by on mountain tops and if it is available very expensive.
So we are looking for other alternatives, ie Wifi links, spread spectrum T1,
etc to have a private network among sites with a single public IP address,
like a home network.

Thanks,
Monty
KE7JVX
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From hkwilliamson at gmail.com  Thu Feb  4 21:21:28 2010
From: hkwilliamson at gmail.com (Keith Williamson)
Date: Thu, 4 Feb 2010 14:21:28 -0700
Subject: [App_rpt-users] Private Network
In-Reply-To: <14bc5ca01002022328v78d59f88gfc218aeb467f2cf8@mail.gmail.com>
References: <14bc5ca01002011240n3c613a3dyda0d0005b81f6c66@mail.gmail.com>
	<14bc5ca01002022328v78d59f88gfc218aeb467f2cf8@mail.gmail.com>
Message-ID: <d15872231002041321m55fd9f31nde357e8936bc3799@mail.gmail.com>

Hi Monty,

It's definitely problematic. You could tie a pair of servers together
if one is Allstar and the other is vanilla Asterisk like for PBX use.
In the router that provides the NAT service for the two servers, you
could setup port forwarding for IAX and Echolink to the Allstar node
and port forwarding for SIP (i.e. 5060) to the PBX server. Not sure if
you might still have issues with RTP forwarding though since SIP needs
it and IAX might need it too. Hmmm.. I suppose you could try using
non-standard ports for IAX, Echolink, SIP, etc but am not sure if you
can get those services to listen on alternate ports (I'm not in front
of my node currently to check).

Necessity is the mother of invention though and it would be an
interesting exercise to attempt. If you do succeed in making it work
for a pair of Allstar nodes you can make it work for any number

73's neighbor!

Keith

KF7DRV

On 2/3/10, Monty Dana <monty at ke7jvx.com> wrote:
> Hello Everyone!
>
> I wondering how one might set up a private network of connected nodes?
> Right now I have two nodes setup here at my home, and would like to
> experiment with having them both tied together on my private home network,
> and still have one node able to access allstar and echolink, is that
> possible?
>
> What the main reason for this question is that many of you know, broadband
> is hard to come by on mountain tops and if it is available very expensive.
> So we are looking for other alternatives, ie Wifi links, spread spectrum T1,
> etc to have a private network among sites with a single public IP address,
> like a home network.
>
> Thanks,
> Monty
> KE7JVX
>

-- 
Sent from my mobile device


From KF6AAQ at gmx.com  Thu Feb  4 21:59:35 2010
From: KF6AAQ at gmx.com (Clint Frost)
Date: Thu, 04 Feb 2010 15:59:35 -0600
Subject: [App_rpt-users] Private Network
In-Reply-To: <d15872231002041321m55fd9f31nde357e8936bc3799@mail.gmail.com>
References: <14bc5ca01002011240n3c613a3dyda0d0005b81f6c66@mail.gmail.com>
	<14bc5ca01002022328v78d59f88gfc218aeb467f2cf8@mail.gmail.com>
	<d15872231002041321m55fd9f31nde357e8936bc3799@mail.gmail.com>
Message-ID: <1265320775.3736.12.camel@cf-52-laptop>

Hello Monty,

I'm not big into networking but would this work for you?

Option 1

Some ISP's will let you have multiple IP address into your home. Once
the public IP's are in your home you should be able to assign them to
each of your Asterisk servers. If the servers are remote then use your
(ham wifi links etc.) 

Yes it would be more expensive to get more public IP's, but it would be
cheaper then having an ISP drop at the hill top :-)

Option 2

This is more of a question or a possible option. I see folks using cell
phones (Storm Chasers on discovery channel) to push live video via
mobile cell phone connection. Would it be possible to use a high speed
3g/4g cell connection on the hill top for your Internet feed? If so that
would be much cheaper then (long range ham wifi links etc)... 

But who wants to rely on cell phones !?!?!?!?!?!!!!!!!!!!

73,

Clint KF6AAQ
NODE 27058 (OFF LINE)



On Thu, 2010-02-04 at 14:21 -0700, Keith Williamson wrote:

> Hi Monty,
> 
> It's definitely problematic. You could tie a pair of servers together
> if one is Allstar and the other is vanilla Asterisk like for PBX use.
> In the router that provides the NAT service for the two servers, you
> could setup port forwarding for IAX and Echolink to the Allstar node
> and port forwarding for SIP (i.e. 5060) to the PBX server. Not sure if
> you might still have issues with RTP forwarding though since SIP needs
> it and IAX might need it too. Hmmm.. I suppose you could try using
> non-standard ports for IAX, Echolink, SIP, etc but am not sure if you
> can get those services to listen on alternate ports (I'm not in front
> of my node currently to check).
> 
> Necessity is the mother of invention though and it would be an
> interesting exercise to attempt. If you do succeed in making it work
> for a pair of Allstar nodes you can make it work for any number
> 
> 73's neighbor!
> 
> Keith
> 
> KF7DRV
> 
> On 2/3/10, Monty Dana <monty at ke7jvx.com> wrote:
> > Hello Everyone!
> >
> > I wondering how one might set up a private network of connected nodes?
> > Right now I have two nodes setup here at my home, and would like to
> > experiment with having them both tied together on my private home network,
> > and still have one node able to access allstar and echolink, is that
> > possible?
> >
> > What the main reason for this question is that many of you know, broadband
> > is hard to come by on mountain tops and if it is available very expensive.
> > So we are looking for other alternatives, ie Wifi links, spread spectrum T1,
> > etc to have a private network among sites with a single public IP address,
> > like a home network.
> >
> > Thanks,
> > Monty
> > KE7JVX
> >
> 




-- 
___________________________________________________________________
Clint Frost
Harvest, Alabama U.S.A.
(256) 542-1223
kf6aaq at gmx.com
Skype: KF6AAQ

To invent, you need a good imagination & a pile of junk.
 ? Thomas Edison  
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From ckraly at gmail.com  Sun Feb  7 15:21:34 2010
From: ckraly at gmail.com (Chuck Kraly)
Date: Sun, 7 Feb 2010 09:21:34 -0600
Subject: [App_rpt-users] Problems with Cepstral and Festival
Message-ID: <16c865551002070721u49bda3d5s8faee0b993781a63@mail.gmail.com>

Hi Guys,

I have been trying to get Cepstral and Festival to both or either play with
ACID. The installs go well, and I can execue them from CLI.
The pc hesitates while it executes, but I get no sound what so ever out of
either the USB fob, or local sound connection. If somoen on the list has
time, could you email me at ckraly at gmail.com and help me see what I might be
missing?


Thanks
Chuck K0XM
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From hkwilliamson at gmail.com  Sun Feb  7 19:29:03 2010
From: hkwilliamson at gmail.com (Keith Williamson)
Date: Sun, 7 Feb 2010 12:29:03 -0700
Subject: [App_rpt-users] Calling into the node operator's local SIP phone
Message-ID: <d15872231002071129s1ee7eb51p9dbe5cd48aef7853@mail.gmail.com>

Hi,

Several people have expressed an interest in how to configure Allstar to
allow radio users to connect to the node-operator's local SIP phone. It
turns out it's pretty easy (once you have a local SIP phone configured of
course). To do this without configuring more general autopatch access with
it's potential risks, you can create a specific context for just allowing a
radio user to access one "outbound" SIP connection, the local SIP phone. In
rpt.conf, I uncommented the "autopatchdn" function and duplicated and
uncommented the "autopatchup" function. These two functions are in the
[functions] stanza available to radio users. In the new "autopatchup"
function, I changed the default DTMF string from 6 to 61 and added the
"context=" option. I set the option to "context=node-op". So the function
now looks like this:

61=autopatchup,context=node-op,noct=1,farenddisconnect=1,dialtime=20000

Then in extensions.conf, I added a stanza for [node-op]:

[node-op]
exten => 1,1,Answer
exten => 1,n,Dial(SIP/200,10)
exten => 1,n,Playback(vm-nobodyavail)
exten => 1,n,Hangup

Change the SIP/200 above to SIP/whatever-your-local-extension-is. Since we
modified rpt.conf, you need to restart asterisk.

Now, if the radio user keys in *611, the autopatch will be invoked and
extension "1" in context [node-op] will be called where it will be answered
and then will dial the local SIP phone extension. If you don't pickup, it
will timeout, play the "nobody available to take your call" message, and
hangup.

Of course you can integrate this into a full autopatch configuration by
modifying the default dialplan in context "radio" but I'm not willing to
open my node up to dialing out to the world (yet).

73's,

Keith
KF7DRV
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From hkwilliamson at gmail.com  Sun Feb  7 21:28:26 2010
From: hkwilliamson at gmail.com (Keith Williamson)
Date: Sun, 7 Feb 2010 14:28:26 -0700
Subject: [App_rpt-users] Calling into the node operator's local SIP phone
In-Reply-To: <8390870f1002071253x5e15532br7374400be0899c8e@mail.gmail.com>
References: <d15872231002071129s1ee7eb51p9dbe5cd48aef7853@mail.gmail.com>
	<8390870f1002071253x5e15532br7374400be0899c8e@mail.gmail.com>
Message-ID: <d15872231002071328r58921fbid6fc6f31cc14daf0@mail.gmail.com>

On Sun, Feb 7, 2010 at 1:53 PM, Stephen - K1LNX <k1lnx at k1lnx.net> wrote:

> Keith,
>     This works like a champ! I just so happened to re-flash one of my Cisco
> phones back to SIP this afternoon for playing with and the timing could not
> have been more perfect lol. A very very useful feature!
>
> 73
> Stephen
> K1LNX
>
>
> On Sun, Feb 7, 2010 at 2:29 PM, Keith Williamson <hkwilliamson at gmail.com>wrote:
>
>> Hi,
>>
>> Several people have expressed an interest in how to configure Allstar to
>> allow radio users to connect to the node-operator's local SIP phone. It
>> turns out it's pretty easy (once you have a local SIP phone configured of
>> course). To do this without configuring more general autopatch access with
>> it's potential risks, you can create a specific context for just allowing a
>> radio user to access one "outbound" SIP connection, the local SIP phone. In
>> rpt.conf, I uncommented the "autopatchdn" function and duplicated and
>> uncommented the "autopatchup" function. These two functions are in the
>> [functions] stanza available to radio users. In the new "autopatchup"
>> function, I changed the default DTMF string from 6 to 61 and added the
>> "context=" option. I set the option to "context=node-op". So the function
>> now looks like this:
>>
>> 61=autopatchup,context=node-op,noct=1,farenddisconnect=1,dialtime=20000
>>
>> Then in extensions.conf, I added a stanza for [node-op]:
>>
>> [node-op]
>> exten => 1,1,Answer
>> exten => 1,n,Dial(SIP/200,10)
>> exten => 1,n,Playback(vm-nobodyavail)
>> exten => 1,n,Hangup
>>
>> Change the SIP/200 above to SIP/whatever-your-local-extension-is. Since we
>> modified rpt.conf, you need to restart asterisk.
>>
>> Now, if the radio user keys in *611, the autopatch will be invoked and
>> extension "1" in context [node-op] will be called where it will be answered
>> and then will dial the local SIP phone extension. If you don't pickup, it
>> will timeout, play the "nobody available to take your call" message, and
>> hangup.
>>
>> Of course you can integrate this into a full autopatch configuration by
>> modifying the default dialplan in context "radio" but I'm not willing to
>> open my node up to dialing out to the world (yet).
>>
>> 73's,
>>
>> Keith
>> KF7DRV
>>
>>
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at qrvc.com
>> http://qrvc.com/mailman/listinfo/app_rpt-users
>>
>>
>
>
> --
> **********************************
> Stephen Brown - ARS K1LNX
> Johnson City, TN EM86
> http://www.k1lnx.net
> google voice: 423-665-9367
> **********************************
>

Yeah, isn't allstar/asterisk the greatest? It's like Lego's for computers,
radios, and telephones.

Cheers,

Keith
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From w9drr.ham at gmail.com  Mon Feb  8 15:00:36 2010
From: w9drr.ham at gmail.com (Don Russell)
Date: Mon, 8 Feb 2010 09:00:36 -0600
Subject: [App_rpt-users] Calling into the node operator's local SIP phone
In-Reply-To: <d15872231002071129s1ee7eb51p9dbe5cd48aef7853@mail.gmail.com>
References: <d15872231002071129s1ee7eb51p9dbe5cd48aef7853@mail.gmail.com>
Message-ID: <dbaef2591002080700u4fb1bedex9e5e3b1dd9a51758@mail.gmail.com>

I was thinking more like erector-set. Solid metal, instead of plastic!

There is no other way of doing some of the stuff I have my box doing.
Asterisk and Linux with that magic app_rpt and the chan_usb just making
life possible for my repeater. It wouldn't exist without those tools.

I don't think it would be possible with IRLP, especially because of
their "rules". My operation was called "illegal" by the IRLP guys
and I got kick/banned a couple times because I came in via my
sip phone instead of rf.

I can't even begin to describe the handiness of being able to use a wifi sip
phone and get into my repeater, connect it via echolink or allstar and
have a converstation with someone on the other end in RF.

Linking repeaters, sip/iax phones in/out, reverse patch, autopatch, cepstral
voice for an unlimited number of things, remote base, shell scripts...
leaves us only to the limit of our imagination.

Thanks again Jim, the Steves', Mark, Scott,  and anyone else who has had
their hands
in the code.

--
Don Russell, CBRE
Director of IT/Chief Operator - Maverick Media
W9DRR - ARRL OES, Technical Specialist
Winnebago County AEC
http://www.socialengineer.us
--
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From rhammock.hur at gmail.com  Mon Feb  8 23:05:44 2010
From: rhammock.hur at gmail.com (Randy Hammock)
Date: Mon, 8 Feb 2010 15:05:44 -0800
Subject: [App_rpt-users] individual graphical node status
Message-ID: <d7c52bb41002081505n2640f400n77e7c23112f6f98a@mail.gmail.com>

Is there a way to get an individual Graphical Node Status Page? For
example, for a local web page, to show a "Zoloft" diagram ( as Jim
calls it) for the nodes connected to 2035 without the rest of the
world.

-- 
Randy Hammock KC6HUR
http://kc6hur.net
IRLP 4494 / ALLSTAR 2035
http://irlp.kc6hur.net


From sales at qrvc.com  Tue Feb  9 01:29:08 2010
From: sales at qrvc.com (Stephen Rodgers)
Date: Mon, 08 Feb 2010 17:29:08 -0800
Subject: [App_rpt-users] individual graphical node status
In-Reply-To: <d7c52bb41002081505n2640f400n77e7c23112f6f98a@mail.gmail.com>
References: <d7c52bb41002081505n2640f400n77e7c23112f6f98a@mail.gmail.com>
Message-ID: <4B70BA64.8010106@qrvc.com>

Randy Hammock wrote:
> Is there a way to get an individual Graphical Node Status Page? For
> example, for a local web page, to show a "Zoloft" diagram ( as Jim
> calls it) for the nodes connected to 2035 without the rest of the
> world.
> 
Not at this time.

Steve
WA6ZFT



From ke2n at cs.com  Tue Feb  9 14:44:12 2010
From: ke2n at cs.com (Ken)
Date: Tue, 09 Feb 2010 09:44:12 -0500
Subject: [App_rpt-users] 64 bit
In-Reply-To: <8CC773670C22941-27FC-3923@webmail-m024.sysops.aol.com>
References: <8CC773670C22941-27FC-3923@webmail-m024.sysops.aol.com>
Message-ID: <8CC77AD7A73FAE7-2008-14B@webmail-m024.sysops.aol.com>

I know that CentOS and Asterisk are available in 64-bit versions
I have a 64 bit machine.
Is it possible to make an install of app_rpt that runs in 64 bit?
if so, how?
Using the standard ISO and install procedure, it installs 32 bit OS and 
modules.

Thanks
Ken Jamrogowicz





From steve at michiganbroadband.com  Tue Feb  9 16:38:03 2010
From: steve at michiganbroadband.com (Steve Gladden)
Date: Tue, 9 Feb 2010 11:38:03 -0500 (EST)
Subject: [App_rpt-users] 64 bit
In-Reply-To: <8CC77AD7A73FAE7-2008-14B@webmail-m024.sysops.aol.com>
References: <8CC773670C22941-27FC-3923@webmail-m024.sysops.aol.com>
	<8CC77AD7A73FAE7-2008-14B@webmail-m024.sysops.aol.com>
Message-ID: <49399.98.250.154.100.1265733483.squirrel@iridium7.michiganbroadband.com>

Hello Ken,
I've done it in the past by copying all of the scripts from
a live 32 bit system over to the 64 bit system.
I first did all of the Centos updates on the 64 bit system..

Then I ran the stage2 script on the 64bit system..

It worked a few months ago..

I've (not by choice_) had to step away from this fun project for a few
months to focus on a new client.

I can't wait to get back into the game.

-Steve





> I know that CentOS and Asterisk are available in 64-bit versions
> I have a 64 bit machine.
> Is it possible to make an install of app_rpt that runs in 64 bit?
> if so, how?
> Using the standard ISO and install procedure, it installs 32 bit OS and
> modules.
>
> Thanks
> Ken Jamrogowicz
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
>


Michigan Broadband Systems Inc.
"Always Connected"

(734)527-7150

Steve's cellphone: (734)904-1811


-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.



From ke2n at cs.com  Tue Feb  9 18:21:49 2010
From: ke2n at cs.com (Ken)
Date: Tue, 9 Feb 2010 13:21:49 -0500
Subject: [App_rpt-users] 64 bit
In-Reply-To: <49399.98.250.154.100.1265733483.squirrel@iridium7.michiganbroadband.com>
References: <8CC773670C22941-27FC-3923@webmail-m024.sysops.aol.com>
	<8CC77AD7A73FAE7-2008-14B@webmail-m024.sysops.aol.com>
	<49399.98.250.154.100.1265733483.squirrel@iridium7.michiganbroadband.com>
Message-ID: <000d01caa9b4$bef846f0$3ce8d4d0$@com>

Thanks Steve - see my question below

> -----Original Message-----
> From: Steve Gladden [mailto:steve at michiganbroadband.com]
> Sent: Tuesday, February 09, 2010 11:38 AM
> To: Ken
> Cc: app_rpt-users at qrvc.com
> Subject: Re: [App_rpt-users] 64 bit
> 
> Hello Ken,
> I've done it in the past by copying all of the scripts from
> a live 32 bit system over to the 64 bit system.
> I first did all of the Centos updates on the 64 bit system..
> 
> Then I ran the stage2 script on the 64bit system..

=====
I would probably need more help with this than you could provide-
But, can you clarify - does this result in a "normal" 32 bit APP_RPT running
under the 64 bit OS?

Or does it use the 64 bit compiler, creating a 64-bit version of the
"forked" Asterisk code?
(This latter one sounds like it has some risk of not working).

Ken Jamrogowicz



From steve at michiganbroadband.com  Tue Feb  9 19:59:58 2010
From: steve at michiganbroadband.com (Steve Gladden)
Date: Tue, 9 Feb 2010 14:59:58 -0500 (EST)
Subject: [App_rpt-users] 64 bit
In-Reply-To: <000d01caa9b4$bef846f0$3ce8d4d0$@com>
References: <8CC773670C22941-27FC-3923@webmail-m024.sysops.aol.com>
	<8CC77AD7A73FAE7-2008-14B@webmail-m024.sysops.aol.com>
	<49399.98.250.154.100.1265733483.squirrel@iridium7.michiganbroadband.com>
	<000d01caa9b4$bef846f0$3ce8d4d0$@com>
Message-ID: <49582.98.250.154.100.1265745598.squirrel@iridium7.michiganbroadband.com>




OOOoooohhhh..

Excellent question!
I'd like to know that as well.
:-)

Steve


> Thanks Steve - see my question below
>
>> -----Original Message-----
>> From: Steve Gladden [mailto:steve at michiganbroadband.com]
>> Sent: Tuesday, February 09, 2010 11:38 AM
>> To: Ken
>> Cc: app_rpt-users at qrvc.com
>> Subject: Re: [App_rpt-users] 64 bit
>>
>> Hello Ken,
>> I've done it in the past by copying all of the scripts from
>> a live 32 bit system over to the 64 bit system.
>> I first did all of the Centos updates on the 64 bit system..
>>
>> Then I ran the stage2 script on the 64bit system..
>
> =====
> I would probably need more help with this than you could provide-
> But, can you clarify - does this result in a "normal" 32 bit APP_RPT
> running
> under the 64 bit OS?
>
> Or does it use the 64 bit compiler, creating a 64-bit version of the
> "forked" Asterisk code?
> (This latter one sounds like it has some risk of not working).
>
> Ken Jamrogowicz
>
>
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
>


Michigan Broadband Systems Inc.
"Always Connected"

(734)527-7150

Steve's cellphone: (734)904-1811


-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.



From hwstar at rodgers.sdcoxmail.com  Tue Feb  9 20:42:33 2010
From: hwstar at rodgers.sdcoxmail.com (hwstar at rodgers.sdcoxmail.com)
Date: Tue, 9 Feb 2010 12:42:33 -0800
Subject: [App_rpt-users] 64 bit
Message-ID: <20100209204233.MHQE18514.dukecmfep06.coxmail.com@dukecmimpo02.coxmail.com>


> 
> From: "Ken" <ke2n at cs.com>
> Date: 2010/02/09 Tue AM 10:21:49 PST
> To: "'Steve Gladden'" <steve at michiganbroadband.com>
> CC: app_rpt-users at qrvc.com
> Subject: Re: [App_rpt-users] 64 bit
> 
> Thanks Steve - see my question below
> 
> > -----Original Message-----
> > From: Steve Gladden [mailto:steve at michiganbroadband.com]
> > Sent: Tuesday, February 09, 2010 11:38 AM
> > To: Ken
> > Cc: app_rpt-users at qrvc.com
> > Subject: Re: [App_rpt-users] 64 bit
> > 
> > Hello Ken,
> > I've done it in the past by copying all of the scripts from
> > a live 32 bit system over to the 64 bit system.
> > I first did all of the Centos updates on the 64 bit system..
> > 
> > Then I ran the stage2 script on the 64bit system..
> 
> =====
> I would probably need more help with this than you could provide-
> But, can you clarify - does this result in a "normal" 32 bit APP_RPT running
> under the 64 bit OS?
> 
> Or does it use the 64 bit compiler, creating a 64-bit version of the
> "forked" Asterisk code?
> (This latter one sounds like it has some risk of not working).
> 
> Ken Jamrogowicz
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
> 

You'll get 64 bit code of the forked version it it compiles without errors. It may work or it may not work I don't know. You would be on your own and we would not be able to support such a build as it is outside of the scope of the project.

Steve
WA6ZFT






From w9drr.ham at gmail.com  Tue Feb  9 22:09:49 2010
From: w9drr.ham at gmail.com (Don Russell)
Date: Tue, 9 Feb 2010 16:09:49 -0600
Subject: [App_rpt-users] 64 bit
In-Reply-To: <20100209204233.MHQE18514.dukecmfep06.coxmail.com@dukecmimpo02.coxmail.com>
References: <20100209204233.MHQE18514.dukecmfep06.coxmail.com@dukecmimpo02.coxmail.com>
Message-ID: <dbaef2591002091409r6fde930nd70a13686d0635c@mail.gmail.com>

If you compile 32 bit code on a 64 bit machine don't you end up with 48 bit
binaries?

Oh, come on guys, thats funny right there....

--
Don Russell, CBRE
Director of IT/Chief Operator - Maverick Media
W9DRR - ARRL OES, Technical Specialist
Winnebago County AEC
http://www.socialengineer.us
--


On Tue, Feb 9, 2010 at 14:42, <hwstar at rodgers.sdcoxmail.com> wrote:

>
> >
> > From: "Ken" <ke2n at cs.com>
> > Date: 2010/02/09 Tue AM 10:21:49 PST
> > To: "'Steve Gladden'" <steve at michiganbroadband.com>
> > CC: app_rpt-users at qrvc.com
> > Subject: Re: [App_rpt-users] 64 bit
> >
> > Thanks Steve - see my question below
> >
> > > -----Original Message-----
> > > From: Steve Gladden [mailto:steve at michiganbroadband.com]
> > > Sent: Tuesday, February 09, 2010 11:38 AM
> > > To: Ken
> > > Cc: app_rpt-users at qrvc.com
> > > Subject: Re: [App_rpt-users] 64 bit
> > >
> > > Hello Ken,
> > > I've done it in the past by copying all of the scripts from
> > > a live 32 bit system over to the 64 bit system.
> > > I first did all of the Centos updates on the 64 bit system..
> > >
> > > Then I ran the stage2 script on the 64bit system..
> >
> > =====
> > I would probably need more help with this than you could provide-
> > But, can you clarify - does this result in a "normal" 32 bit APP_RPT
> running
> > under the 64 bit OS?
> >
> > Or does it use the 64 bit compiler, creating a 64-bit version of the
> > "forked" Asterisk code?
> > (This latter one sounds like it has some risk of not working).
> >
> > Ken Jamrogowicz
> >
> > _______________________________________________
> > App_rpt-users mailing list
> > App_rpt-users at qrvc.com
> > http://qrvc.com/mailman/listinfo/app_rpt-users
> >
>
> You'll get 64 bit code of the forked version it it compiles without errors.
> It may work or it may not work I don't know. You would be on your own and we
> would not be able to support such a build as it is outside of the scope of
> the project.
>
> Steve
> WA6ZFT
>
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
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From kb2ear at kb2ear.net  Wed Feb 10 20:28:03 2010
From: kb2ear at kb2ear.net (Scott Weis)
Date: Wed, 10 Feb 2010 15:28:03 -0500
Subject: [App_rpt-users] 64 bit
In-Reply-To: <20100209204233.MHQE18514.dukecmfep06.coxmail.com@dukecmimpo02.coxmail.com>
References: <20100209204233.MHQE18514.dukecmfep06.coxmail.com@dukecmimpo02.coxmail.com>
Message-ID: <4EC5BF13F9354D71A4A06C5FEB18640B@KB2EAR2>

Just for fun I installed a 64-bit Centos on a machine and ran the phase1.sh 
script etc. When it completed I had a fully 64 bit version up and running 
with URI.

73 de
Scott KB2EAR
----- Original Message ----- 
From: <hwstar at rodgers.sdcoxmail.com>
To: "Ken" <ke2n at cs.com>; "'Steve Gladden'" <steve at michiganbroadband.com>
Cc: <app_rpt-users at qrvc.com>
Sent: Tuesday, February 09, 2010 3:42 PM
Subject: Re: [App_rpt-users] 64 bit


>
>>
>> From: "Ken" <ke2n at cs.com>
>> Date: 2010/02/09 Tue AM 10:21:49 PST
>> To: "'Steve Gladden'" <steve at michiganbroadband.com>
>> CC: app_rpt-users at qrvc.com
>> Subject: Re: [App_rpt-users] 64 bit
>>
>> Thanks Steve - see my question below
>>
>> > -----Original Message-----
>> > From: Steve Gladden [mailto:steve at michiganbroadband.com]
>> > Sent: Tuesday, February 09, 2010 11:38 AM
>> > To: Ken
>> > Cc: app_rpt-users at qrvc.com
>> > Subject: Re: [App_rpt-users] 64 bit
>> >
>> > Hello Ken,
>> > I've done it in the past by copying all of the scripts from
>> > a live 32 bit system over to the 64 bit system.
>> > I first did all of the Centos updates on the 64 bit system..
>> >
>> > Then I ran the stage2 script on the 64bit system..
>>
>> =====
>> I would probably need more help with this than you could provide-
>> But, can you clarify - does this result in a "normal" 32 bit APP_RPT 
>> running
>> under the 64 bit OS?
>>
>> Or does it use the 64 bit compiler, creating a 64-bit version of the
>> "forked" Asterisk code?
>> (This latter one sounds like it has some risk of not working).
>>
>> Ken Jamrogowicz
>>
>> _______________________________________________
>> App_rpt-users mailing list
>> App_rpt-users at qrvc.com
>> http://qrvc.com/mailman/listinfo/app_rpt-users
>>
>
> You'll get 64 bit code of the forked version it it compiles without 
> errors. It may work or it may not work I don't know. You would be on your 
> own and we would not be able to support such a build as it is outside of 
> the scope of the project.
>
> Steve
> WA6ZFT
>
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
> 




From hwstar at rodgers.sdcoxmail.com  Wed Feb 10 22:38:11 2010
From: hwstar at rodgers.sdcoxmail.com (hwstar at rodgers.sdcoxmail.com)
Date: Wed, 10 Feb 2010 14:38:11 -0800
Subject: [App_rpt-users] 64 bit
Message-ID: <20100210223811.ZXCK19302.dukecmfep05.coxmail.com@dukecmimpo01.coxmail.com>


> 
> From: "Scott Weis" <kb2ear at kb2ear.net>
> Date: 2010/02/10 Wed PM 12:28:03 PST
> To: app_rpt-users at qrvc.com
> Subject: Re: [App_rpt-users] 64 bit
> 
> Just for fun I installed a 64-bit Centos on a machine and ran the phase1.sh 
> script etc. When it completed I had a fully 64 bit version up and running 
> with URI.
> 
> 73 de
> Scott KB2EAR
> ----- Original Message ----- 
> From: <hwstar at rodgers.sdcoxmail.com>
> To: "Ken" <ke2n at cs.com>; "'Steve Gladden'" <steve at michiganbroadband.com>
> Cc: <app_rpt-users at qrvc.com>
> Sent: Tuesday, February 09, 2010 3:42 PM
> Subject: Re: [App_rpt-users] 64 bit
> 
> 
> >
> >>
> >> From: "Ken" <ke2n at cs.com>
> >> Date: 2010/02/09 Tue AM 10:21:49 PST
> >> To: "'Steve Gladden'" <steve at michiganbroadband.com>
> >> CC: app_rpt-users at qrvc.com
> >> Subject: Re: [App_rpt-users] 64 bit
> >>
> >> Thanks Steve - see my question below
> >>
> >> > -----Original Message-----
> >> > From: Steve Gladden [mailto:steve at michiganbroadband.com]
> >> > Sent: Tuesday, February 09, 2010 11:38 AM
> >> > To: Ken
> >> > Cc: app_rpt-users at qrvc.com
> >> > Subject: Re: [App_rpt-users] 64 bit
> >> >
> >> > Hello Ken,
> >> > I've done it in the past by copying all of the scripts from
> >> > a live 32 bit system over to the 64 bit system.
> >> > I first did all of the Centos updates on the 64 bit system..
> >> >
> >> > Then I ran the stage2 script on the 64bit system..
> >>
> >> =====
> >> I would probably need more help with this than you could provide-
> >> But, can you clarify - does this result in a "normal" 32 bit APP_RPT 
> >> running
> >> under the 64 bit OS?
> >>
> >> Or does it use the 64 bit compiler, creating a 64-bit version of the
> >> "forked" Asterisk code?
> >> (This latter one sounds like it has some risk of not working).
> >>
> >> Ken Jamrogowicz
> >>
> >> _______________________________________________
> >> App_rpt-users mailing list
> >> App_rpt-users at qrvc.com
> >> http://qrvc.com/mailman/listinfo/app_rpt-users
> >>
> >
> > You'll get 64 bit code of the forked version it it compiles without 
> > errors. It may work or it may not work I don't know. You would be on your 
> > own and we would not be able to support such a build as it is outside of 
> > the scope of the project.
> >
> > Steve
> > WA6ZFT
> >
> >
> >
> >
> > _______________________________________________
> > App_rpt-users mailing list
> > App_rpt-users at qrvc.com
> > http://qrvc.com/mailman/listinfo/app_rpt-users
> >
> > 
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
> 

That may appear to be true, but without testing every last feature and configuration you won't know for sure. I'm hesitant to call something good when it was built and only briefly tested. However, if enough people test their specific configurations and have a positive experience then maybe it is mostly good. 

Steve
WA6ZFT




From g4rky at yahoo.co.uk  Fri Feb 12 14:30:28 2010
From: g4rky at yahoo.co.uk (Matt Beasant)
Date: Fri, 12 Feb 2010 14:30:28 +0000
Subject: [App_rpt-users] Connpgm / disconpgm use
Message-ID: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>

Hi all!

I've been following some discussions on the IRLP list about node owners
setting up their nodes to send node status updates to Twitter.

I was wondering if I could do something similar with my app_rpt node? Has
anyone already done this?

I'm guessing I'll need to use the connpgm and disconpgm entries in rpt.conf.

How do I deal with the arguments that get passed out with those commands?
What are they called?

The IRLP script uses curl to send info to twitter, would this work with
App_rpt too?

Sorry for the dumb noob questions!

Thanks,

Matt
G4RKY
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From g4rky at yahoo.co.uk  Mon Feb 15 18:18:32 2010
From: g4rky at yahoo.co.uk (Matt Beasant)
Date: Mon, 15 Feb 2010 18:18:32 +0000
Subject: [App_rpt-users] Connpgm / disconpgm use
In-Reply-To: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>
References: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>
Message-ID: <ff95e2ea1002151018o5d23ed23xc4229a0fe09d0e4c@mail.gmail.com>

OK well that was a deafening silence! I guess I am in uncharted territory
here..

I have managed to get a "tweet" for connect and disconnect states of my node
using curl so, so far so good.

What I dont know and cant seem to work out from the source code, is the
names of the variables that are passed as arguments. I tried using $name and
$them as I thought that might work but no luck.

If anyone can help identify the variables that are available for use by an
external script, that would be fantastic, so thanks in advance.

app_rpt is clearly nothing like IRLP when it comes to variables and
stuff......

73,

Matt

On 12 February 2010 14:30, Matt Beasant <g4rky at yahoo.co.uk> wrote:

> Hi all!
>
> I've been following some discussions on the IRLP list about node owners
> setting up their nodes to send node status updates to Twitter.
>
> I was wondering if I could do something similar with my app_rpt node? Has
> anyone already done this?
>
> I'm guessing I'll need to use the connpgm and disconpgm entries in
> rpt.conf.
>
> How do I deal with the arguments that get passed out with those commands?
> What are they called?
>
> The IRLP script uses curl to send info to twitter, would this work with
> App_rpt too?
>
> Sorry for the dumb noob questions!
>
> Thanks,
>
> Matt
> G4RKY
>
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From telesistant at hotmail.com  Mon Feb 15 19:03:31 2010
From: telesistant at hotmail.com (Jim Duuuude)
Date: Mon, 15 Feb 2010 11:03:31 -0800
Subject: [App_rpt-users] New features / bug fixes
In-Reply-To: <ff95e2ea1002151018o5d23ed23xc4229a0fe09d0e4c@mail.gmail.com>
References: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>,
	<ff95e2ea1002151018o5d23ed23xc4229a0fe09d0e4c@mail.gmail.com>
Message-ID: <BAY119-W1799913F908AD4F25DA64CB14A0@phx.gbl>



The following Mantis issues are now resolved:

22 -- LiTZ mode
40 -- Cannot go into carrier squelch mode without commenting out the rxctcssfreqs and txctcssfreqs options.
41 -- Archivedir recording feature only records node TX audio and not Rx audio
49 -- Function class to execute a command or shell script via DTMF

This requires upgrade to current apps/app_rpt.c (version 0.211),
channels/chan_usbradio.c and main/dsp.c.

Sorry that I have been out of it for a while, but I have been having some
significant health problems and have been doing the best I possibly can.

P.S.

Does anyone have a test system running IRLP that is loaded with the latest ACID
distribution so that I can look at and resolve one of the bugs easily??  Thanks.


JIM WB6NIL


 		 	   		  
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From matt.g4rky at googlemail.com  Mon Feb 15 19:10:47 2010
From: matt.g4rky at googlemail.com (Matt Beasant)
Date: Mon, 15 Feb 2010 19:10:47 +0000
Subject: [App_rpt-users] New features / bug fixes
In-Reply-To: <BAY119-W1799913F908AD4F25DA64CB14A0@phx.gbl>
References: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>
	<ff95e2ea1002151018o5d23ed23xc4229a0fe09d0e4c@mail.gmail.com>
	<BAY119-W1799913F908AD4F25DA64CB14A0@phx.gbl>
Message-ID: <ff95e2ea1002151110v227e776fk3a3e44f611fca0c@mail.gmail.com>

Sure Jim, you're welcome to use my node - I have Echolink and IRLP running
on 2250.

Only too happy to help.

( Is this the jumping audio level on IRLP issue? Cos I still get that one )

Email me off list and I'll give you the SSH details.

Matt
G4RKY

On 15 February 2010 19:03, Jim Duuuude <telesistant at hotmail.com> wrote:

>
> The following Mantis issues are now resolved:
>
> 22 -- LiTZ mode
> 40 -- Cannot go into carrier squelch mode without commenting out the
> rxctcssfreqs and txctcssfreqs options.
> 41 -- Archivedir recording feature only records node TX audio and not Rx
> audio
> 49 -- Function class to execute a command or shell script via DTMF
>
> This requires upgrade to current apps/app_rpt.c (version 0.211),
> channels/chan_usbradio.c and main/dsp.c.
>
> Sorry that I have been out of it for a while, but I have been having some
> significant health problems and have been doing the best I possibly can.
>
> P.S.
>
> Does anyone have a test system running IRLP that is loaded with the latest
> ACID
> distribution so that I can look at and resolve one of the bugs easily??
> Thanks.
>
>
> JIM WB6NIL
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
>
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From g4rky at yahoo.co.uk  Mon Feb 15 19:26:38 2010
From: g4rky at yahoo.co.uk (Matt Beasant)
Date: Mon, 15 Feb 2010 19:26:38 +0000
Subject: [App_rpt-users] Connpgm / disconpgm use
In-Reply-To: <BAY119-W233C35C8DB474D90E9B8C8B14A0@phx.gbl>
References: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>
	<ff95e2ea1002151018o5d23ed23xc4229a0fe09d0e4c@mail.gmail.com>
	<BAY119-W233C35C8DB474D90E9B8C8B14A0@phx.gbl>
Message-ID: <ff95e2ea1002151126p2041c973xb3c542978001eec3@mail.gmail.com>

On 15 February 2010 18:51, Jim Duuuude <telesistant at hotmail.com> wrote:

>  there are no variables from app_rpt itself other then the node number
>
>
Hi Jim, thanks for the reply.

Sorry about the duff info, I was merely basing my assumption on the info in
the example rpt.conf which states that there are 2 arguments passed to the
connpgm / disconpgm :

connpgm=/etc/connectprog ; . Execute a program you specify on connect.
; passes 2 command line arguments to your program:

I was wondering how to use these arguments so I can include them in my
tweet...
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From g4rky at yahoo.co.uk  Mon Feb 15 22:08:01 2010
From: g4rky at yahoo.co.uk (Matt Beasant)
Date: Mon, 15 Feb 2010 22:08:01 +0000
Subject: [App_rpt-users] Tweeting node status
Message-ID: <ff95e2ea1002151408y3d361daex309da876d04f0adb@mail.gmail.com>

I've cracked it, if anyone else is interested in this then email me direct!

Cheers,

Matt

On 12 February 2010 14:30, Matt Beasant <g4rky at yahoo.co.uk> wrote:

> Hi all!
>
> I've been following some discussions on the IRLP list about node owners
> setting up their nodes to send node status updates to Twitter.
>
> I was wondering if I could do something similar with my app_rpt node? Has
> anyone already done this?
>
> I'm guessing I'll need to use the connpgm and disconpgm entries in
> rpt.conf.
>
> How do I deal with the arguments that get passed out with those commands?
> What are they called?
>
> The IRLP script uses curl to send info to twitter, would this work with
> App_rpt too?
>
> Sorry for the dumb noob questions!
>
> Thanks,
>
> Matt
> G4RKY
>
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From telesistant at hotmail.com  Tue Feb 16 03:37:01 2010
From: telesistant at hotmail.com (Jim Duuuude)
Date: Mon, 15 Feb 2010 19:37:01 -0800
Subject: [App_rpt-users] New features / bug fixes
In-Reply-To: <BAY119-W1799913F908AD4F25DA64CB14A0@phx.gbl>
References: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>, ,
	<ff95e2ea1002151018o5d23ed23xc4229a0fe09d0e4c@mail.gmail.com>,
	<BAY119-W1799913F908AD4F25DA64CB14A0@phx.gbl>
Message-ID: <BAY119-W295837AE442526A1574646B1490@phx.gbl>


I implemented a couple more today:

42 -- 
		Add the ability to use a COP command to enable and disable receive CTCSS
48 -- TX CTCSS only on valid input
54 -- 
		nighly rsync updates aren't running

This requires upgrade to current apps/app_rpt.c (version 0.213),
channels/chan_usbradio.c, channels/xpmr/xpmr.c and channels/xpmr/xpmr.h.

JIM WB6NIL

From: telesistant at hotmail.com
To: app_rpt-users at qrvc.com
Date: Mon, 15 Feb 2010 11:03:31 -0800
Subject: [App_rpt-users] New features / bug fixes









The following Mantis issues are now resolved:

22 -- LiTZ mode
40 -- Cannot go into carrier squelch mode without commenting out the rxctcssfreqs and txctcssfreqs options.
41 -- Archivedir recording feature only records node TX audio and not Rx audio
49 -- Function class to execute a command or shell script via DTMF

This requires upgrade to current apps/app_rpt.c (version 0.211),
channels/chan_usbradio.c and main/dsp.c.

Sorry that I have been out of it for a while, but I have been having some
significant health problems and have been doing the best I possibly can.

P.S.

Does anyone have a test system running IRLP that is loaded with the latest ACID
distribution so that I can look at and resolve one of the bugs easily??  Thanks.


JIM WB6NIL


 		 	   		  
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From pjgjunk at comcast.net  Tue Feb 16 13:09:05 2010
From: pjgjunk at comcast.net (pjgjunk at comcast.net)
Date: Tue, 16 Feb 2010 08:09:05 -0500
Subject: [App_rpt-users] USB Controller requirment
Message-ID: <3DECCA03D15D456C99DB166DF4D0E913@dell1505>

Hi all,

 

I thought I had read that USB 2.0 was required for the USB FOB, but I cannot
fine it anywhere in the documentation.  Both systems I have showed that the
FOB is running at 12 MHz.

 

Should this be running at 480 MHz?

 

# ./usbtree.pl

/: Bus 04.Port 1: Dev 1, Class=root_hub, Drv=uhci_hcd/2p, 12M

/: Bus 03.Port 1: Dev 1, Class=root_hub, Drv=ohci_hcd/2p, 12M

    |_ Port 2: Dev 2, If 0, Prod=C-Media USB Headphone Set, Class=audio,
Drv=snd-usb-audio, 12M

    |_ Port 2: Dev 2, If 1, Prod=, Class=audio, Drv=snd-usb-audio, 12M

    |_ Port 2: Dev 2, If 2, Prod=, Class=audio, Drv=snd-usb-audio, 12M

    |_ Port 2: Dev 2, If 3, Prod=, Class=HID, Drv=usbfs, 12M

/: Bus 02.Port 1: Dev 1, Class=root_hub, Drv=ohci_hcd/3p, 12M

/: Bus 01.Port 1: Dev 1, Class=root_hub, Drv=ehci_hcd/5p, 480M

    |_ Port 2: Dev 2, If 0, Prod=USB DISK 2.0, Class=stor., Drv=usb-storage,
480M

 

I connected a storage device just to prove that the EHCI was functioning.

 

Thanks,

Paul

 

 

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From ke2n at cs.com  Tue Feb 16 13:58:27 2010
From: ke2n at cs.com (Ken)
Date: Tue, 16 Feb 2010 08:58:27 -0500
Subject: [App_rpt-users]  USB Controller requirment
In-Reply-To: <8CC7D26F84064A0-46F8-CF88@webmail-d072.sysops.aol.com>
References: <8CC7D26F84064A0-46F8-CF88@webmail-d072.sysops.aol.com>
Message-ID: <8CC7D273F58F19C-46F8-CFFC@webmail-d072.sysops.aol.com>

Subject: Re: [App_rpt-users] USB Controller requirment


The CM108AH data sheet says it supports the "full speed mode" of USB 2.0.
The full-speed mode is 12 Mb/s
?
The USB 2.0 specification requires hubs to support high-speed mode. USB 2.0 devices are not required to
support high-speed mode.? 
?
There is some app_rpt documentation saying any HUB you use should support 480 Mb/s? - no surprise since this is required by the USB2 spec.

?
Ken
?



-----Original Message-----
From: pjgjunk at comcast.net
To: 'app_rpt mailing list' &lt;app_rpt-users at qrvc.com&gt;
Sent: Tue, Feb 16, 2010 8:09 am
Subject: [App_rpt-users] USB Controller requirment

Hi all,
?
I thought I had read that USB 2.0 was required for the USB FOB, but I cannot fine it anywhere in the documentation.? Both systems I have showed that the FOB is running at 12 MHz.
?
Should this be running at 480 MHz?
?
# ./usbtree.pl
/: Bus 04.Port 1: Dev 1, Class=root_hub, Drv=uhci_hcd/2p, 12M
/: Bus 03.Port 1: Dev 1, Class=root_hub, Drv=ohci_hcd/2p, 12M
??? |_ Port 2: Dev 2, If 0, Prod=C-Media USB Headphone Set, Class=audio, Drv=snd-usb-audio, 12M
??? |_ Port 2: Dev 2, If 1, Prod=, Class=audio, Drv=snd-usb-audio, 12M
??? |_ Port 2: Dev 2, If 2, Prod=, Class=audio, Drv=snd-usb-audio, 12M
??? |_ Port 2: Dev 2, If 3, Prod=, Class=HID, Drv=usbfs, 12M
/: Bus 02.Port 1: Dev 1, Class=root_hub, Drv=ohci_hcd/3p, 12M
/: Bus 01.Port 1: Dev 1, Class=root_hub, Drv=ehci_hcd/5p, 480M
??? |_ Port 2: Dev 2, If 0, Prod=USB DISK 2.0, Class=stor., Drv=usb-storage, 480M
?
I connected a storage device just to prove that the EHCI was functioning.
?
Thanks,
Paul
?
?


_______________________________________________ App_rpt-users mailing list App_rpt-users at qrvc.com http://qrvc.com/mailman/listinfo/app_rpt-users 

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From k1lnx at k1lnx.net  Wed Feb 17 01:26:00 2010
From: k1lnx at k1lnx.net (Stephen - K1LNX)
Date: Tue, 16 Feb 2010 20:26:00 -0500
Subject: [App_rpt-users] LiTZ Usage
Message-ID: <8390870f1002161726q4d288c30pde781a16155e5acf@mail.gmail.com>

How is the new LiTZ feature implemented? I've updated to the latest code in
the repository, but I don't see any examples in the config files, unless I
looked over it?

I'd like to start experimenting with this, I've got a SIP phone hanging off
of my app_rpt box that I would like to make ring in response to a LiTZ event
for example.

Anyone have any config examples?

tnx and 73
Stephen
K1LNX

-- 
****************************************
Stephen Brown - ARS K1LNX
Johnson City, TN EM86
http://www.k1lnx.net
google voice: 423-665-9367
sip: sbrown at voip.stephennet.net
****************************************
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From k1lnx at k1lnx.net  Wed Feb 17 01:53:50 2010
From: k1lnx at k1lnx.net (Stephen - K1LNX)
Date: Tue, 16 Feb 2010 20:53:50 -0500
Subject: [App_rpt-users] LiTZ Usage
In-Reply-To: <8390870f1002161726q4d288c30pde781a16155e5acf@mail.gmail.com>
References: <8390870f1002161726q4d288c30pde781a16155e5acf@mail.gmail.com>
Message-ID: <8390870f1002161753o615fb5aem517c111748f327a1@mail.gmail.com>

An update, this was easier than I thought, I dug through the source a bit
and figured it out I think.

Add litzcmd=dosomethinghere to your node stanza in rpt.conf

For example in mine, I have it set to ring my SIP phone with *62:
litzcmd=*62    ; Long Tone Zero, ring node operator's SIP phone

You should see something similar to this in the Asterisk console:
[Feb 16 20:49:30] NOTICE[18459]: chan_usbradio.c:2440 usbradio_read: Got
DTMF char 0 duration 3455 ms
[Feb 16 20:49:30] NOTICE[18459]: app_rpt.c:16933 rpt: Doing litz command *62
on node 2335

Very slick, I could see a million different ways to implement this!!!

Stephen
K1LNX

On Tue, Feb 16, 2010 at 8:26 PM, Stephen - K1LNX <k1lnx at k1lnx.net> wrote:

> How is the new LiTZ feature implemented? I've updated to the latest code in
> the repository, but I don't see any examples in the config files, unless I
> looked over it?
>
> I'd like to start experimenting with this, I've got a SIP phone hanging off
> of my app_rpt box that I would like to make ring in response to a LiTZ event
> for example.
>
> Anyone have any config examples?
>
> tnx and 73
> Stephen
> K1LNX
>
> --
> ****************************************
> Stephen Brown - ARS K1LNX
> Johnson City, TN EM86
> http://www.k1lnx.net
> google voice: 423-665-9367
> sip: sbrown at voip.stephennet.net
> ****************************************
>



-- 
****************************************
Stephen Brown - ARS K1LNX
Johnson City, TN EM86
http://www.k1lnx.net
google voice: 423-665-9367
sip: sbrown at voip.stephennet.net
****************************************
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From nessenj at jimsoffice.org  Wed Feb 17 02:07:30 2010
From: nessenj at jimsoffice.org (James Nessen)
Date: Tue, 16 Feb 2010 18:07:30 -0800
Subject: [App_rpt-users] LiTZ Usage
In-Reply-To: <8390870f1002161753o615fb5aem517c111748f327a1@mail.gmail.com>
References: <8390870f1002161726q4d288c30pde781a16155e5acf@mail.gmail.com>
	<8390870f1002161753o615fb5aem517c111748f327a1@mail.gmail.com>
Message-ID: <91a3416d1002161807u743c0802w6daa5d6a4a152c9a@mail.gmail.com>

You can also find a write up here:

http://app-rpt.qrvc.com/node/171

Jim, K6JWN


On Tue, Feb 16, 2010 at 5:53 PM, Stephen - K1LNX <k1lnx at k1lnx.net> wrote:

> An update, this was easier than I thought, I dug through the source a bit
> and figured it out I think.
>
> Add litzcmd=dosomethinghere to your node stanza in rpt.conf
>
> For example in mine, I have it set to ring my SIP phone with *62:
> litzcmd=*62    ; Long Tone Zero, ring node operator's SIP phone
>
> You should see something similar to this in the Asterisk console:
> [Feb 16 20:49:30] NOTICE[18459]: chan_usbradio.c:2440 usbradio_read: Got
> DTMF char 0 duration 3455 ms
> [Feb 16 20:49:30] NOTICE[18459]: app_rpt.c:16933 rpt: Doing litz command
> *62 on node 2335
>
> Very slick, I could see a million different ways to implement this!!!
>
> Stephen
> K1LNX
>
>
> On Tue, Feb 16, 2010 at 8:26 PM, Stephen - K1LNX <k1lnx at k1lnx.net> wrote:
>
>> How is the new LiTZ feature implemented? I've updated to the latest code
>> in the repository, but I don't see any examples in the config files, unless
>> I looked over it?
>>
>> I'd like to start experimenting with this, I've got a SIP phone hanging
>> off of my app_rpt box that I would like to make ring in response to a LiTZ
>> event for example.
>>
>> Anyone have any config examples?
>>
>> tnx and 73
>> Stephen
>> K1LNX
>>
>> --
>> ****************************************
>> Stephen Brown - ARS K1LNX
>> Johnson City, TN EM86
>> http://www.k1lnx.net
>> google voice: 423-665-9367
>> sip: sbrown at voip.stephennet.net
>> ****************************************
>>
>
>
>
> --
> ****************************************
> Stephen Brown - ARS K1LNX
> Johnson City, TN EM86
> http://www.k1lnx.net
> google voice: 423-665-9367
> sip: sbrown at voip.stephennet.net
> ****************************************
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
>


-- 
Jim Nessen K6JWN
Email: nessenj at jimsoffice.org | Ph: 530.564.0039
IRLP: Nodes 3598,7358 | Echolink: 83598 K6JWN-R
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From telesistant at hotmail.com  Wed Feb 17 16:59:20 2010
From: telesistant at hotmail.com (Jim Duuuude)
Date: Wed, 17 Feb 2010 08:59:20 -0800
Subject: [App_rpt-users] New features / bug fixes
In-Reply-To: <358e96271002170828v49ac8420i7b36813f0aa0e1dc@mail.gmail.com>
References: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>,
	<ff95e2ea1002151018o5d23ed23xc4229a0fe09d0e4c@mail.gmail.com>,
	<BAY119-W1799913F908AD4F25DA64CB14A0@phx.gbl>,
	<BAY119-W295837AE442526A1574646B1490@phx.gbl>,
	<358e96271002170828v49ac8420i7b36813f0aa0e1dc@mail.gmail.com>
Message-ID: <BAY119-W15B8A5DEF5846209166FF5B1480@phx.gbl>


Note, the lines without '#' are the ones that are typed into the Linux
prompt:


cd /usr/src
#or whatever directory has the 'asterisk' directory which contains
#the asterisk, zaptel and libri sources are.

mv asterisk asterisk.old 
#this saves the old source tree as 'asterisk.old'

svn co svn://qrvc.com/projects/allstar/astsrc-1.4.23-pre/trunk
#this downloads the new source tree from SVN

mv trunk asterisk
#this renames the newly downloaded source tree as 'asterisk'

cd asterisk
cd zaptel
make install
cd ../libpri
make install
cd ../asterisk
./configure
make install
#Re-builds and installs the whole thing



JIM 


Date: Wed, 17 Feb 2010 10:28:35 -0600
Subject: Re: [App_rpt-users] New features / bug fixes
From: bbrown at byrg.net
To: telesistant at hotmail.com

hey man
?
i have had my head in sand ... how do i update from ver .210 to .213?
?
i did the: ?astupd.sh
and it says i have latest..... (.210)
?
am i missing a proceedure here?
--
Thanks in Advance

? Bob Brown, W?NQX

? Kansas City Metro Area

? http://sm0kenet.net

? http://byrg.net


? http://kcdstar.byrg.net

Quis custodiet ipsos custodes?
?



On Mon, Feb 15, 2010 at 21:37, Jim Duuuude <telesistant at hotmail.com> wrote:


I implemented a couple more today:

42 -- Add the ability to use a COP command to enable and disable receive CTCSS
48 -- TX CTCSS only on valid input
54 -- nighly rsync updates aren't running


This requires upgrade to current apps/app_rpt.c (version 0.213),
channels/chan_usbradio.c, channels/xpmr/xpmr.c and channels/xpmr/xpmr.h.

JIM WB6NIL




From: telesistant at hotmail.com
To: app_rpt-users at qrvc.com
Date: Mon, 15 Feb 2010 11:03:31 -0800

Subject: [App_rpt-users] New features / bug fixes





The following Mantis issues are now resolved:

22 -- LiTZ mode
40 -- Cannot go into carrier squelch mode without commenting out the rxctcssfreqs and txctcssfreqs options.
41 -- Archivedir recording feature only records node TX audio and not Rx audio

49 -- Function class to execute a command or shell script via DTMF

This requires upgrade to current apps/app_rpt.c (version 0.211),
channels/chan_usbradio.c and main/dsp.c.

Sorry that I have been out of it for a while, but I have been having some

significant health problems and have been doing the best I possibly can.

P.S.

Does anyone have a test system running IRLP that is loaded with the latest ACID
distribution so that I can look at and resolve one of the bugs easily??? Thanks.



JIM WB6NIL



_______________________________________________
App_rpt-users mailing list
App_rpt-users at qrvc.com
http://qrvc.com/mailman/listinfo/app_rpt-users



 		 	   		  

From w9drr.ham at gmail.com  Wed Feb 17 18:10:37 2010
From: w9drr.ham at gmail.com (Don Russell)
Date: Wed, 17 Feb 2010 12:10:37 -0600
Subject: [App_rpt-users] New features / bug fixes
In-Reply-To: <BAY119-W15B8A5DEF5846209166FF5B1480@phx.gbl>
References: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>
	<ff95e2ea1002151018o5d23ed23xc4229a0fe09d0e4c@mail.gmail.com>
	<BAY119-W1799913F908AD4F25DA64CB14A0@phx.gbl>
	<BAY119-W295837AE442526A1574646B1490@phx.gbl>
	<358e96271002170828v49ac8420i7b36813f0aa0e1dc@mail.gmail.com>
	<BAY119-W15B8A5DEF5846209166FF5B1480@phx.gbl>
Message-ID: <dbaef2591002171010o18f90545q67d85a2d0418ffb8@mail.gmail.com>

I had an issue compiling zaptel with kernel 2.6.30 (gentoo). ztdummy
wouldn't compile.
Known issue and there is a patch available. I attached the patch which
should probably
be merged into the tree. I was going to put it in mantis but didn't see the
proper spot.


--
Don Russell, CBRE
Director of IT/Chief Operator - Maverick Media
W9DRR - ARRL OES, Technical Specialist
Winnebago County AEC
http://www.socialengineer.us
--


On Wed, Feb 17, 2010 at 10:59, Jim Duuuude <telesistant at hotmail.com> wrote:

>
> Note, the lines without '#' are the ones that are typed into the Linux
> prompt:
>
>
> cd /usr/src
> #or whatever directory has the 'asterisk' directory which contains
> #the asterisk, zaptel and libri sources are.
>
> mv asterisk asterisk.old
> #this saves the old source tree as 'asterisk.old'
>
> svn co svn://qrvc.com/projects/allstar/astsrc-1.4.23-pre/trunk
> #this downloads the new source tree from SVN
>
> mv trunk asterisk
> #this renames the newly downloaded source tree as 'asterisk'
>
> cd asterisk
> cd zaptel
> make install
> cd ../libpri
> make install
> cd ../asterisk
> ./configure
> make install
> #Re-builds and installs the whole thing
>
>
>
> JIM
>
>
> Date: Wed, 17 Feb 2010 10:28:35 -0600
> Subject: Re: [App_rpt-users] New features / bug fixes
> From: bbrown at byrg.net
> To: telesistant at hotmail.com
>
> hey man
>
> i have had my head in sand ... how do i update from ver .210 to .213?
>
> i did the:  astupd.sh
> and it says i have latest..... (.210)
>
> am i missing a proceedure here?
> --
> Thanks in Advance
>
>   Bob Brown, W?NQX
>
>   Kansas City Metro Area
>
>   http://sm0kenet.net
>
>   http://byrg.net
>
>
>   http://kcdstar.byrg.net
>
> Quis custodiet ipsos custodes?
>
>
>
>
> On Mon, Feb 15, 2010 at 21:37, Jim Duuuude <telesistant at hotmail.com>
> wrote:
>
>
> I implemented a couple more today:
>
> 42 -- Add the ability to use a COP command to enable and disable receive
> CTCSS
> 48 -- TX CTCSS only on valid input
> 54 -- nighly rsync updates aren't running
>
>
> This requires upgrade to current apps/app_rpt.c (version 0.213),
> channels/chan_usbradio.c, channels/xpmr/xpmr.c and channels/xpmr/xpmr.h.
>
> JIM WB6NIL
>
>
>
>
> From: telesistant at hotmail.com
> To: app_rpt-users at qrvc.com
> Date: Mon, 15 Feb 2010 11:03:31 -0800
>
> Subject: [App_rpt-users] New features / bug fixes
>
>
>
>
>
> The following Mantis issues are now resolved:
>
> 22 -- LiTZ mode
> 40 -- Cannot go into carrier squelch mode without commenting out the
> rxctcssfreqs and txctcssfreqs options.
> 41 -- Archivedir recording feature only records node TX audio and not Rx
> audio
>
> 49 -- Function class to execute a command or shell script via DTMF
>
> This requires upgrade to current apps/app_rpt.c (version 0.211),
> channels/chan_usbradio.c and main/dsp.c.
>
> Sorry that I have been out of it for a while, but I have been having some
>
> significant health problems and have been doing the best I possibly can.
>
> P.S.
>
> Does anyone have a test system running IRLP that is loaded with the latest
> ACID
> distribution so that I can look at and resolve one of the bugs easily??
> Thanks.
>
>
>
> JIM WB6NIL
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
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From k1lnx at k1lnx.net  Wed Feb 17 18:43:39 2010
From: k1lnx at k1lnx.net (Stephen - K1LNX)
Date: Wed, 17 Feb 2010 13:43:39 -0500
Subject: [App_rpt-users] New features / bug fixes
In-Reply-To: <BAY119-W15B8A5DEF5846209166FF5B1480@phx.gbl>
References: <ff95e2ea1002120630p5c360689q63f69998d048cfc2@mail.gmail.com>
	<ff95e2ea1002151018o5d23ed23xc4229a0fe09d0e4c@mail.gmail.com>
	<BAY119-W1799913F908AD4F25DA64CB14A0@phx.gbl>
	<BAY119-W295837AE442526A1574646B1490@phx.gbl>
	<358e96271002170828v49ac8420i7b36813f0aa0e1dc@mail.gmail.com>
	<BAY119-W15B8A5DEF5846209166FF5B1480@phx.gbl>
Message-ID: <8390870f1002171043u160c5bd0nc353e7aec7737917@mail.gmail.com>

Were there any updates to zaptel or libpri? I only saw a few config files
and Asterisk files come down from SVN against my existing tree. As a result,
I only rebuilt Asterisk. It's working fine, so should I assume that to be
the case?

Stephen
K1LNX

On Wed, Feb 17, 2010 at 11:59 AM, Jim Duuuude <telesistant at hotmail.com>wrote:

>
> Note, the lines without '#' are the ones that are typed into the Linux
> prompt:
>
>
> cd /usr/src
> #or whatever directory has the 'asterisk' directory which contains
> #the asterisk, zaptel and libri sources are.
>
> mv asterisk asterisk.old
> #this saves the old source tree as 'asterisk.old'
>
> svn co svn://qrvc.com/projects/allstar/astsrc-1.4.23-pre/trunk
> #this <http://qrvc.com/projects/allstar/astsrc-1.4.23-pre/trunk%0A#this>downloads the new source tree from SVN
>
> mv trunk asterisk
> #this renames the newly downloaded source tree as 'asterisk'
>
> cd asterisk
> cd zaptel
> make install
> cd ../libpri
> make install
> cd ../asterisk
> ./configure
> make install
> #Re-builds and installs the whole thing
>
>
>
> JIM
>
>
> Date: Wed, 17 Feb 2010 10:28:35 -0600
> Subject: Re: [App_rpt-users] New features / bug fixes
> From: bbrown at byrg.net
> To: telesistant at hotmail.com
>
> hey man
>
> i have had my head in sand ... how do i update from ver .210 to .213?
>
> i did the:  astupd.sh
> and it says i have latest..... (.210)
>
> am i missing a proceedure here?
> --
> Thanks in Advance
>
>   Bob Brown, W?NQX
>
>   Kansas City Metro Area
>
>   http://sm0kenet.net
>
>   http://byrg.net
>
>
>   http://kcdstar.byrg.net
>
> Quis custodiet ipsos custodes?
>
>
>
>
> On Mon, Feb 15, 2010 at 21:37, Jim Duuuude <telesistant at hotmail.com>
> wrote:
>
>
> I implemented a couple more today:
>
> 42 -- Add the ability to use a COP command to enable and disable receive
> CTCSS
> 48 -- TX CTCSS only on valid input
> 54 -- nighly rsync updates aren't running
>
>
> This requires upgrade to current apps/app_rpt.c (version 0.213),
> channels/chan_usbradio.c, channels/xpmr/xpmr.c and channels/xpmr/xpmr.h.
>
> JIM WB6NIL
>
>
>
>
> From: telesistant at hotmail.com
> To: app_rpt-users at qrvc.com
> Date: Mon, 15 Feb 2010 11:03:31 -0800
>
> Subject: [App_rpt-users] New features / bug fixes
>
>
>
>
>
> The following Mantis issues are now resolved:
>
> 22 -- LiTZ mode
> 40 -- Cannot go into carrier squelch mode without commenting out the
> rxctcssfreqs and txctcssfreqs options.
> 41 -- Archivedir recording feature only records node TX audio and not Rx
> audio
>
> 49 -- Function class to execute a command or shell script via DTMF
>
> This requires upgrade to current apps/app_rpt.c (version 0.211),
> channels/chan_usbradio.c and main/dsp.c.
>
> Sorry that I have been out of it for a while, but I have been having some
>
> significant health problems and have been doing the best I possibly can.
>
> P.S.
>
> Does anyone have a test system running IRLP that is loaded with the latest
> ACID
> distribution so that I can look at and resolve one of the bugs easily??
> Thanks.
>
>
>
> JIM WB6NIL
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
>
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>



-- 
****************************************
Stephen Brown - ARS K1LNX
Johnson City, TN EM86
http://www.k1lnx.net
google voice: 423-665-9367
sip: sbrown at voip.stephennet.net
****************************************
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From hwstar at rodgers.sdcoxmail.com  Wed Feb 17 19:37:22 2010
From: hwstar at rodgers.sdcoxmail.com (hwstar at rodgers.sdcoxmail.com)
Date: Wed, 17 Feb 2010 11:37:22 -0800
Subject: [App_rpt-users] New features / bug fixes
Message-ID: <20100217193722.DITZ27309.dukecmfep06.coxmail.com@dukecmimpo03.coxmail.com>


> 
> From: Stephen - K1LNX <k1lnx at k1lnx.net>
> Date: 2010/02/17 Wed AM 10:43:39 PST
> To: app_rpt mailing list <app_rpt-users at qrvc.com>
> Subject: Re: [App_rpt-users] New features / bug fixes
> 
> Were there any updates to zaptel or libpri? I only saw a few config files
> and Asterisk files come down from SVN against my existing tree. As a result,
> I only rebuilt Asterisk. It's working fine, so should I assume that to be
> the case?
> 
> Stephen
> K1LNX
> 
> On Wed, Feb 17, 2010 at 11:59 AM, Jim Duuuude <telesistant at hotmail.com>wrote:
> 
> >
> > Note, the lines without '#' are the ones that are typed into the Linux
> > prompt:
> >
> >
> > cd /usr/src
> > #or whatever directory has the 'asterisk' directory which contains
> > #the asterisk, zaptel and libri sources are.
> >
> > mv asterisk asterisk.old
> > #this saves the old source tree as 'asterisk.old'
> >
> > svn co svn://qrvc.com/projects/allstar/astsrc-1.4.23-pre/trunk
> > #this <http://qrvc.com/projects/allstar/astsrc-1.4.23-pre/trunk%0A#this>downloads the new source tree from SVN
> >
> > mv trunk asterisk
> > #this renames the newly downloaded source tree as 'asterisk'
> >
> > cd asterisk
> > cd zaptel
> > make install
> > cd ../libpri
> > make install
> > cd ../asterisk
> > ./configure
> > make install
> > #Re-builds and installs the whole thing
> >
> >
> >
> > JIM
> >
> >
> > Date: Wed, 17 Feb 2010 10:28:35 -0600
> > Subject: Re: [App_rpt-users] New features / bug fixes
> > From: bbrown at byrg.net
> > To: telesistant at hotmail.com
> >
> > hey man
> >
> > i have had my head in sand ... how do i update from ver .210 to .213?
> >
> > i did the:  astupd.sh
> > and it says i have latest..... (.210)
> >
> > am i missing a proceedure here?
> > --
> > Thanks in Advance
> >
> >   Bob Brown, W?NQX
> >
> >   Kansas City Metro Area
> >
> >   http://sm0kenet.net
> >
> >   http://byrg.net
> >
> >
> >   http://kcdstar.byrg.net
> >
> > Quis custodiet ipsos custodes?
> >
> >
> >
> >
> > On Mon, Feb 15, 2010 at 21:37, Jim Duuuude <telesistant at hotmail.com>
> > wrote:
> >
> >
> > I implemented a couple more today:
> >
> > 42 -- Add the ability to use a COP command to enable and disable receive
> > CTCSS
> > 48 -- TX CTCSS only on valid input
> > 54 -- nighly rsync updates aren't running
> >
> >
> > This requires upgrade to current apps/app_rpt.c (version 0.213),
> > channels/chan_usbradio.c, channels/xpmr/xpmr.c and channels/xpmr/xpmr.h.
> >
> > JIM WB6NIL
> >
> >
> >
> >
> > From: telesistant at hotmail.com
> > To: app_rpt-users at qrvc.com
> > Date: Mon, 15 Feb 2010 11:03:31 -0800
> >
> > Subject: [App_rpt-users] New features / bug fixes
> >
> >
> >
> >
> >
> > The following Mantis issues are now resolved:
> >
> > 22 -- LiTZ mode
> > 40 -- Cannot go into carrier squelch mode without commenting out the
> > rxctcssfreqs and txctcssfreqs options.
> > 41 -- Archivedir recording feature only records node TX audio and not Rx
> > audio
> >
> > 49 -- Function class to execute a command or shell script via DTMF
> >
> > This requires upgrade to current apps/app_rpt.c (version 0.211),
> > channels/chan_usbradio.c and main/dsp.c.
> >
> > Sorry that I have been out of it for a while, but I have been having some
> >
> > significant health problems and have been doing the best I possibly can.
> >
> > P.S.
> >
> > Does anyone have a test system running IRLP that is loaded with the latest
> > ACID
> > distribution so that I can look at and resolve one of the bugs easily??
> > Thanks.
> >
> >
> >
> > JIM WB6NIL
> >
> >
> >
> > _______________________________________________
> > App_rpt-users mailing list
> > App_rpt-users at qrvc.com
> > http://qrvc.com/mailman/listinfo/app_rpt-users
> >
> >
> >
> >
> > _______________________________________________
> > App_rpt-users mailing list
> > App_rpt-users at qrvc.com
> > http://qrvc.com/mailman/listinfo/app_rpt-users
> >
> 
> 
> 
> -- 
> ****************************************
> Stephen Brown - ARS K1LNX
> Johnson City, TN EM86
> http://www.k1lnx.net
> google voice: 423-665-9367
> sip: sbrown at voip.stephennet.net
> ****************************************
> 
> 
You should never make assumptions about what changed and always rebuild all 3 parts that were downloaded by SVN: asterisk, libpri, and zaptel.

Steve
WA6ZFT

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http://qrvc.com/mailman/listinfo/app_rpt-users


From g4rky at yahoo.co.uk  Wed Feb 17 22:21:17 2010
From: g4rky at yahoo.co.uk (Matt Beasant)
Date: Wed, 17 Feb 2010 22:21:17 +0000
Subject: [App_rpt-users] Tweeting node status
In-Reply-To: <ff95e2ea1002151408y3d361daex309da876d04f0adb@mail.gmail.com>
References: <ff95e2ea1002151408y3d361daex309da876d04f0adb@mail.gmail.com>
Message-ID: <ff95e2ea1002171421i15c9da1dmf3a25cc02e40c3aa@mail.gmail.com>

Dear All,

Here is my latest attempt at my connect script which posts the node's status
on Twitter.

It now detects the type of connection ( IRLP, Echolink or Allstar ) and also
has a lookup for some often used connections from my node to put some more
detail in the tweet.

I am sure the Linux gurus out there will cringe at my crappy attempt at
scripting but it works and you gotta start somewhere :-D

I bet there are much better ways of doing this but this is all I could come
up with tonight.

Here it is for those that might want to do something similar:

# Detect if Echolink node number or not
if  [ $2 -lt 300000 ] ; then
let nodeno=($2-300000)
systype="Echolink node "
if [ $2 -eq 3001332 ] ; then
name="MB7IBA"
fi
fi

# Detect if IRLP node number or not
if  [ $2 -lt 49999 ]; then
let nodeno=($2-40000)
systype="IRLP Node "
if  [ $2 -eq 49755 ] ; then
name="UK Reflector"
elif [ $2 -eq 49453 ] ; then
name="WIN System, California"
fi
fi

# Detect if Allstar node number or not
if [ $2 -lt 3000 ]; then
let nodeno=$2
systype="AllStar Node "
if [ $2 -eq 2237 ] ; then
name="GB3ZY"
elif [ $2 -eq 2259 ] ; then
name="GB3DQ"
fi
fi

# Tweet it
curl -u username:password -d status="GB3FH connected  to  $systype $nodeno
$name" http://twitter.com/statuses/update.xml &




On 15 February 2010 22:08, Matt Beasant <g4rky at yahoo.co.uk> wrote:

> I've cracked it, if anyone else is interested in this then email me direct!
>
> Cheers,
>
> Matt
>
> On 12 February 2010 14:30, Matt Beasant <g4rky at yahoo.co.uk> wrote:
>
>> Hi all!
>>
>> I've been following some discussions on the IRLP list about node owners
>> setting up their nodes to send node status updates to Twitter.
>>
>> I was wondering if I could do something similar with my app_rpt node? Has
>> anyone already done this?
>>
>> I'm guessing I'll need to use the connpgm and disconpgm entries in
>> rpt.conf.
>>
>> How do I deal with the arguments that get passed out with those commands?
>> What are they called?
>>
>> The IRLP script uses curl to send info to twitter, would this work with
>> App_rpt too?
>>
>> Sorry for the dumb noob questions!
>>
>> Thanks,
>>
>> Matt
>> G4RKY
>>
>
>
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From ke2n at cs.com  Thu Feb 18 15:16:15 2010
From: ke2n at cs.com (Ken)
Date: Thu, 18 Feb 2010 10:16:15 -0500
Subject: [App_rpt-users]  New features / bug fixes
In-Reply-To: <8390870f1002171043u160c5bd0nc353e7aec7737917@mail.gmail.com>
References: <8390870f1002171043u160c5bd0nc353e7aec7737917@mail.gmail.com>
Message-ID: <8CC7EC472BAA3C4-32A4-1C6E7@webmail-m034.sysops.aol.com>

Is this essentially the same things as
?
astupd.sh


for those with the standard "ACID" installation?
?
Ken Jamrogowicz

On Wed, Feb 17, 2010 at 11:59 AM, Jim Duuuude &lt;telesistant at hotmail.com&gt; wrote:

Note, the lines without '#' are the ones that are typed into the Linux
prompt:


cd /usr/src
#or whatever directory has the 'asterisk' directory which contains
#the asterisk, zaptel and libri sources are.

mv asterisk asterisk.old
#this saves the old source tree as 'asterisk.old'

svn co svn://qrvc.com/projects/allstar/astsrc-1.4.23-pre/trunk
#this downloads the new source tree from SVN

mv trunk asterisk
#this renames the newly downloaded source tree as 'asterisk'

cd asterisk
cd zaptel
make install
cd ../libpri
make install
cd ../asterisk
./configure
make install
#Re-builds and installs the whole thing

JIM

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From sales at qrvc.com  Thu Feb 18 15:31:18 2010
From: sales at qrvc.com (Stephen Rodgers)
Date: Thu, 18 Feb 2010 07:31:18 -0800
Subject: [App_rpt-users] New features / bug fixes
In-Reply-To: <8CC7EC472BAA3C4-32A4-1C6E7@webmail-m034.sysops.aol.com>
References: <8390870f1002171043u160c5bd0nc353e7aec7737917@mail.gmail.com>
	<8CC7EC472BAA3C4-32A4-1C6E7@webmail-m034.sysops.aol.com>
Message-ID: <4B7D5D46.4090303@qrvc.com>

Ken wrote:
> Is this essentially the same things as
> ?
> astupd.sh
> 
> 
> for those with the standard "ACID" installation?
> ?
> Ken Jamrogowicz
> 
> On Wed, Feb 17, 2010 at 11:59 AM, Jim Duuuude &lt;telesistant at hotmail.com&gt; wrote:
> 
> Note, the lines without '#' are the ones that are typed into the Linux
> prompt:
> 
> 
> cd /usr/src
> #or whatever directory has the 'asterisk' directory which contains
> #the asterisk, zaptel and libri sources are.
> 
> mv asterisk asterisk.old
> #this saves the old source tree as 'asterisk.old'
> 
> svn co svn://qrvc.com/projects/allstar/astsrc-1.4.23-pre/trunk
> #this downloads the new source tree from SVN
> 
> mv trunk asterisk
> #this renames the newly downloaded source tree as 'asterisk'
> 
> cd asterisk
> cd zaptel
> make install
> cd ../libpri
> make install
> cd ../asterisk
> ./configure
> make install
> #Re-builds and installs the whole thing
> 
> JIM
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users

The changes are in the source tree. but not in the ACID tarball yet.

The astupd script will probably be removed in future releases as it really doesn't accomplish
what I hoped it would do. Any changes to ACID scripts force a version change of ACID and this prevents
astupd from downloading future code updates. Code updates should be done per the instructions on
app_rpt.qrvc.com in the developers section, or using Jim Dixon's method in a prior post.

Steve
WA6ZFT



From cemergen at gmail.com  Thu Feb 18 16:11:47 2010
From: cemergen at gmail.com (Cem ERGEN)
Date: Thu, 18 Feb 2010 18:11:47 +0200
Subject: [App_rpt-users] please help
Message-ID: <9e288e091002180811vfa28764ue2c995895823270b@mail.gmail.com>

hello,

i have installed system elastix on machine1 and it is working fine.. i could
add sip peers and sip peers could contact other sip peears thats ok
also i've motorola sx900 radio ( not only one.. i've many motorola radio)
i would like to connect sip peers to motorola sx900 or inverse...
for example motorola sx900 name is radio1
sip peer name is sip1

i've softphone and login sip1 on softphone then dial radio1 and connect then
talk.

could you help me please how could i do that

-- 
U Can
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From hwstar at rodgers.sdcoxmail.com  Thu Feb 18 19:26:20 2010
From: hwstar at rodgers.sdcoxmail.com (hwstar at rodgers.sdcoxmail.com)
Date: Thu, 18 Feb 2010 11:26:20 -0800
Subject: [App_rpt-users] please help
Message-ID: <20100218192620.NBJU27309.dukecmfep06.coxmail.com@dukecmimpo02.coxmail.com>


> 
> From: Cem ERGEN <cemergen at gmail.com>
> Date: 2010/02/18 Thu AM 08:11:47 PST
> To: app_rpt-users at qrvc.com
> Subject: [App_rpt-users] please help
> 
> hello,
> 
> i have installed system elastix on machine1 and it is working fine.. i could
> add sip peers and sip peers could contact other sip peears thats ok
> also i've motorola sx900 radio ( not only one.. i've many motorola radio)
> i would like to connect sip peers to motorola sx900 or inverse...
> for example motorola sx900 name is radio1
> sip peer name is sip1
> 
> i've softphone and login sip1 on softphone then dial radio1 and connect then
> talk.
> 
> could you help me please how could i do that
> 
> -- 
> U Can
> 
> 

Hello,

You are going to have to explain in more detail the computer system you have, what asterisk version you are running (i.e. is it ACID or Limey, or some custom setup)  and what interface card you are using. Also be prepared to attach configuration files rpt.conf, extensions.conf, iax.conf and sip.conf.

If this is a commercial application requiring support it is best handled privately. Please note that commercial support regarding desired system configurations and or code changes is probably not going to be offered free of charge.


Steve
WA6ZFT

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http://qrvc.com/mailman/listinfo/app_rpt-users


From w4wwm at knology.net  Fri Feb 19 02:22:35 2010
From: w4wwm at knology.net (Will Wright)
Date: Thu, 18 Feb 2010 20:22:35 -0600
Subject: [App_rpt-users] Bring Asterisk back to life
Message-ID: <4B7DF5EB.7040102@knology.net>

Hello to the group,

Need a little help. I call myself running an update for the latest 
ACID.  Once completed Asterisk seen to crash on me with this message in 
a loop. >> Asterisk died with code 1, Automatically restarting Asterisk, 
mpg123: no process kill.

What can I do to bring Asterisk back from the dead?  Thanks

Will / W4WWM



From kb2ear at kb2ear.net  Fri Feb 19 04:57:14 2010
From: kb2ear at kb2ear.net (Scott Weis)
Date: Thu, 18 Feb 2010 23:57:14 -0500
Subject: [App_rpt-users] Bring Asterisk back to life
In-Reply-To: <4B7DF5EB.7040102@knology.net>
References: <4B7DF5EB.7040102@knology.net>
Message-ID: <FAC0BE820AAD445BACA155D994EEEE1C@KB2EAR2>

Mine does that when the URI is disconnected.

Scott
----- Original Message ----- 
From: "Will Wright" <w4wwm at knology.net>
To: <app_rpt-users at qrvc.com>
Sent: Thursday, February 18, 2010 9:22 PM
Subject: [App_rpt-users] Bring Asterisk back to life


> Hello to the group,
> 
> Need a little help. I call myself running an update for the latest 
> ACID.  Once completed Asterisk seen to crash on me with this message in 
> a loop. >> Asterisk died with code 1, Automatically restarting Asterisk, 
> mpg123: no process kill.
> 
> What can I do to bring Asterisk back from the dead?  Thanks
> 
> Will / W4WWM
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
> 
>



From rodrigo at suaconta.com.br  Fri Feb 19 13:06:35 2010
From: rodrigo at suaconta.com.br (Rodrigo)
Date: Fri, 19 Feb 2010 11:06:35 -0200
Subject: [App_rpt-users] app_rpt + PCI QUAD RADIO INTERFACE + asterisk
Message-ID: <4B7E8CDB.2070905@suaconta.com.br>

Hi everybody, my name is Rodrigo and iam a new in app_rpt.

First, sorry about my english, cause is not good.

Can you say where i can get samples of asterisk configuration files 
using pci quad radio interface?
And anybody can giveme some help about dahdi driver to this interface, 
because i have some ploblem to install digium TDM400P with pci quad 
radio interface in asterisk server (ubuntu 9.10 server).

Thank you

Rodrigo de Oliveira S?


From sales at qrvc.com  Fri Feb 19 15:37:37 2010
From: sales at qrvc.com (Stephen Rodgers)
Date: Fri, 19 Feb 2010 07:37:37 -0800
Subject: [App_rpt-users] app_rpt + PCI QUAD RADIO INTERFACE + asterisk
In-Reply-To: <4B7E8CDB.2070905@suaconta.com.br>
References: <4B7E8CDB.2070905@suaconta.com.br>
Message-ID: <4B7EB041.3090202@qrvc.com>

Rodrigo wrote:
> Hi everybody, my name is Rodrigo and iam a new in app_rpt.
> 
> First, sorry about my english, cause is not good.
> 
> Can you say where i can get samples of asterisk configuration files 
> using pci quad radio interface?
> And anybody can giveme some help about dahdi driver to this interface, 
> because i have some ploblem to install digium TDM400P with pci quad 
> radio interface in asterisk server (ubuntu 9.10 server).
> 
> Thank you
> 
> Rodrigo de Oliveira S?
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
> 
The sample config files for the quad radio pci card are available in svn here:

http://qrvc.com/viewvc/projects/allstar/astsrc-1.4.23-pre/trunk/configs/pciradio/?root=svn

There is also a wealth of information about how to configure app_rpt on our website here:

http://app-rpt.qrvc.com/node/59

Steve



From nessenj at jimsoffice.org  Fri Feb 19 20:29:01 2010
From: nessenj at jimsoffice.org (James Nessen)
Date: Fri, 19 Feb 2010 12:29:01 -0800
Subject: [App_rpt-users] chan_echolink?
Message-ID: <91a3416d1002191229hdb470f6pe978ffb92ea29f1d@mail.gmail.com>

is anyone running chan_echolink and having any issues?  I just fired up my
version (/* Version 0.21, 07/09/2008) today and noticed that it's not
downloading the updated user list (I have had my echolink client on my
windows machine running for over 15 minutes and it's not showing up in my
database (via database showkey k6jwn).

I am running ACID with the latest app_rpt (213).  I checked on another node
(running limey) and it seems to be working properly (and updating the
echolink database around every 4 minutes).

Can anyone shed light?

Jim
-- 
Jim Nessen K6JWN
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From kb2ear at kb2ear.net  Sat Feb 20 16:57:16 2010
From: kb2ear at kb2ear.net (Scott Weis)
Date: Sat, 20 Feb 2010 11:57:16 -0500
Subject: [App_rpt-users] USB Hub
Message-ID: <0CD7436AD88945D19A57F6D320819FE3@KB2EAR2>

After trying every USB 2.0 Hub I can find in my office and at home, I have 
found none that work with my setup. Can anyone point me to a good source of 
know working multi-tt hubs?

Thanks & 73
Scott KB2EAR 




From telesistant at hotmail.com  Sat Feb 20 16:59:36 2010
From: telesistant at hotmail.com (Jim Duuuude)
Date: Sat, 20 Feb 2010 08:59:36 -0800
Subject: [App_rpt-users] USB Hub
In-Reply-To: <0CD7436AD88945D19A57F6D320819FE3@KB2EAR2>
References: <0CD7436AD88945D19A57F6D320819FE3@KB2EAR2>
Message-ID: <BAY119-W50480B6C80570C1ABC3E09B1450@phx.gbl>


haven't you heard that friends don't let friends use USB hubs??? :-)

----------------------------------------
> From: kb2ear at kb2ear.net
> To: app_rpt-users at qrvc.com
> Date: Sat, 20 Feb 2010 11:57:16 -0500
> Subject: [App_rpt-users] USB Hub
>
> After trying every USB 2.0 Hub I can find in my office and at home, I have
> found none that work with my setup. Can anyone point me to a good source of
> know working multi-tt hubs?
>
> Thanks & 73
> Scott KB2EAR
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
 		 	   		  

From ke2n at cs.com  Sat Feb 20 18:32:01 2010
From: ke2n at cs.com (Ken)
Date: Sat, 20 Feb 2010 13:32:01 -0500
Subject: [App_rpt-users] USB Hub
In-Reply-To: <BAY119-W50480B6C80570C1ABC3E09B1450@phx.gbl>
References: <BAY119-W50480B6C80570C1ABC3E09B1450@phx.gbl>
Message-ID: <8CC807220D1359F-5020-168@webmail-m096.sysops.aol.com>

tee hee
it's asking for trouble alright - but in my case I simply cannot find a 
PCI card with USB ports that work with Asterisk (usb_radio)
They work fine with everything else except asterisk, for some reason. I 
tried 4 kinds.
These are in a PCI-X slot of either a Dell Power Edge 2650 or a 2850.
This slot is designed for massive throughput like big SCSI arrays or 
fiber channel comms.
But it cannot handle 8K audio for some reason.
If somebody has a card that works in one of these I would like to know 
how they did it

Meanwhile, I have had good luck with a 4-port hub by a Hong Kong 
company called Highstar.

an example can be found on ebay at item number 290398030370

I recommend the orange.

GL

Ken



-----Original Message-----
From: Jim Duuuude <telesistant at hotmail.com>
To: kb2ear at kb2ear.net; app_rpt mailing list <app_rpt-users at qrvc.com>
Sent: Sat, Feb 20, 2010 11:59 am
Subject: Re: [App_rpt-users] USB Hub


haven't you heard that friends don't let friends use USB hubs??? :-)

----------------------------------------
> From: kb2ear at kb2ear.net
> To: app_rpt-users at qrvc.com
> Date: Sat, 20 Feb 2010 11:57:16 -0500
> Subject: [App_rpt-users] USB Hub
>
> After trying every USB 2.0 Hub I can find in my office and at home, I 
have
> found none that work with my setup. Can anyone point me to a good 
source of
> know working multi-tt hubs?
>
> Thanks & 73
> Scott KB2EAR
>
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users

_______________________________________________
App_rpt-users mailing list
App_rpt-users at qrvc.com
http://qrvc.com/mailman/listinfo/app_rpt-users




From ke6sth at ke6sth.ampr.org  Sun Feb 21 07:58:47 2010
From: ke6sth at ke6sth.ampr.org (ke6sth at ke6sth.ampr.org)
Date: Sat, 20 Feb 2010 23:58:47 -0800 (PST)
Subject: [App_rpt-users] chan_echolink does not announce disconnects
In-Reply-To: <8CC807220D1359F-5020-168@webmail-m096.sysops.aol.com>
References: <BAY119-W50480B6C80570C1ABC3E09B1450@phx.gbl>
	<8CC807220D1359F-5020-168@webmail-m096.sysops.aol.com>
Message-ID: <Pine.LNX.4.64.1002202355520.10921@localhost.localdomain>


Hi All,

new here to using my repeater with asterisk. I just got setup using ACID 
everything works greate except one thing, echolink does not announce 
disconnects, is there a setting for that to happen?

Thanks,
--Sione KE6STH



From sales at qrvc.com  Sun Feb 21 16:17:42 2010
From: sales at qrvc.com (Stephen Rodgers)
Date: Sun, 21 Feb 2010 08:17:42 -0800
Subject: [App_rpt-users] chan_echolink does not announce disconnects
In-Reply-To: <Pine.LNX.4.64.1002202355520.10921@localhost.localdomain>
References: <BAY119-W50480B6C80570C1ABC3E09B1450@phx.gbl>	<8CC807220D1359F-5020-168@webmail-m096.sysops.aol.com>
	<Pine.LNX.4.64.1002202355520.10921@localhost.localdomain>
Message-ID: <4B815CA6.8010302@qrvc.com>

ke6sth at ke6sth.ampr.org wrote:
> Hi All,
> 
> new here to using my repeater with asterisk. I just got setup using ACID 
> everything works greate except one thing, echolink does not announce 
> disconnects, is there a setting for that to happen?
> 
> Thanks,
> --Sione KE6STH
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
> 

The default configuration should announce disconnects. If that does not work, please log a bug in Mantis at
mantis.qrvc.com. You'll need to sign up for a reporter account before logging the bug.

Steve
WA6ZFT



From ke2n at cs.com  Sun Feb 21 20:47:46 2010
From: ke2n at cs.com (Ken)
Date: Sun, 21 Feb 2010 15:47:46 -0500
Subject: [App_rpt-users] chan_echolink does not announce disconnects
In-Reply-To: <Pine.LNX.4.64.1002202355520.10921@localhost.localdomain>
References: <Pine.LNX.4.64.1002202355520.10921@localhost.localdomain>
Message-ID: <8CC814E421D3CF8-2DBC-2402E@webmail-d064.sysops.aol.com>

oh - disconnect message for IAX?  I never hear one either, only the 
beep.

here is what appears on the screen

ke2n*CLI> rpt fun 27021 *12560
  == Parsing '/var/lib/asterisk/rpt_extnodes': Found
  == Parsing '/var/lib/asterisk/rpt_extnodes': Found
  == Parsing '/var/lib/asterisk/rpt_extnodes': Found
  == Parsing '/var/lib/asterisk/rpt_extnodes': Found
  == Parsing '/var/lib/asterisk/rpt_extnodes': Found
    -- Hungup 'IAX2/64.27.0.247:4569-15678'
    -- Hungup 'Zap/pseudo-105602502'
    -- Hungup 'Zap/pseudo-1114550565'


   - Ken Jamrogowicz


-----Original Message-----
From: ke6sth at ke6sth.ampr.org
To: app_rpt-users at qrvc.com
Sent: Sun, Feb 21, 2010 2:58 am
Subject: [App_rpt-users] chan_echolink does not announce disconnects


Hi All,

new here to using my repeater with asterisk. I just got setup using 
ACID
everything works greate except one thing, echolink does not announce
disconnects, is there a setting for that to happen?

Thanks,
--Sione KE6STH

_______________________________________________
App_rpt-users mailing list
App_rpt-users at qrvc.com
http://qrvc.com/mailman/listinfo/app_rpt-users




From ke6sth at ke6sth.ampr.org  Sun Feb 21 21:29:44 2010
From: ke6sth at ke6sth.ampr.org (ke6sth at ke6sth.ampr.org)
Date: Sun, 21 Feb 2010 13:29:44 -0800 (PST)
Subject: [App_rpt-users] chan_echolink does not announce disconnects
In-Reply-To: <8CC814E421D3CF8-2DBC-2402E@webmail-d064.sysops.aol.com>
References: <Pine.LNX.4.64.1002202355520.10921@localhost.localdomain>
	<8CC814E421D3CF8-2DBC-2402E@webmail-d064.sysops.aol.com>
Message-ID: <Pine.LNX.4.64.1002211327520.22751@localhost.localdomain>



I don't even hear a beep, is that something I can configure? should it be 
configured by default? I am new at this so this is something I am 
learning at the moment. This is for echolink is that handle using IAX 
channel?

--Sione KE6STH

On Sun, 21 Feb 2010, Ken wrote:

> oh - disconnect message for IAX?  I never hear one either, only the beep.
>
> here is what appears on the screen
>
> ke2n*CLI> rpt fun 27021 *12560
> == Parsing '/var/lib/asterisk/rpt_extnodes': Found
> == Parsing '/var/lib/asterisk/rpt_extnodes': Found
> == Parsing '/var/lib/asterisk/rpt_extnodes': Found
> == Parsing '/var/lib/asterisk/rpt_extnodes': Found
> == Parsing '/var/lib/asterisk/rpt_extnodes': Found
>   -- Hungup 'IAX2/64.27.0.247:4569-15678'
>   -- Hungup 'Zap/pseudo-105602502'
>   -- Hungup 'Zap/pseudo-1114550565'
>
>
>  - Ken Jamrogowicz
>
>
> -----Original Message-----
> From: ke6sth at ke6sth.ampr.org
> To: app_rpt-users at qrvc.com
> Sent: Sun, Feb 21, 2010 2:58 am
> Subject: [App_rpt-users] chan_echolink does not announce disconnects
>
>
> Hi All,
>
> new here to using my repeater with asterisk. I just got setup using ACID
> everything works greate except one thing, echolink does not announce
> disconnects, is there a setting for that to happen?
>
> Thanks,
> --Sione KE6STH
>
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
>
>


From w4wwm at knology.net  Sun Feb 21 22:39:25 2010
From: w4wwm at knology.net (Will Wright)
Date: Sun, 21 Feb 2010 16:39:25 -0600
Subject: [App_rpt-users] Asterisk died with code 1
Message-ID: <4B81B61D.2090700@knology.net>

Steve,

I tried to run the update from version 210 to version 213, but Asterisk 
dies with code 1.

Is there a fix that you can share with me?   I have rebooted to pick up 
one URI,  but I still get the Asterisk died with code 1.  Thanks


Will / W4WWM





From cemergen at gmail.com  Mon Feb 22 09:19:07 2010
From: cemergen at gmail.com (Cem ERGEN)
Date: Mon, 22 Feb 2010 11:19:07 +0200
Subject: [App_rpt-users] hello
Message-ID: <9e288e091002220119t143dfc73p67556c9358716b10@mail.gmail.com>

hi everybody
i've 2 motorola sx900 radio and one dingotel usb 2way adapter
and i want to make a call from sip softphone to motorola sx900 radio.
i've just installed acid on my pc1 and connect dingotel usb adapter then i
see usb  adapter led is flashing.
i suppose it is ready to run. then i connect a cable between motorola sx900
radio ( lets we say radio1) and usb adapter.
radio1 is turned on and it is works. radio2 is turned on and ready.
now i want to make a call sip softphone to radio2

i don't understant and i'm in confuse

i can't add sip peer to acid server how can i do
i can't add radio1 to acid server hi can i do

which number call from my sip phone to connect radio1?

:( sorry my english is poor

could you explain simple and details
thansk a billion





-- 
U Can
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From cemergen at gmail.com  Mon Feb 22 15:46:47 2010
From: cemergen at gmail.com (Cem ERGEN)
Date: Mon, 22 Feb 2010 17:46:47 +0200
Subject: [App_rpt-users] iaxrpt
Message-ID: <9e288e091002220746w754caca0h88f7ad57c70e2cd5@mail.gmail.com>

hello
i have acid server and i'm trying to use iaxrpt program. i added
extensions.conf and iax.conf needed configariton but when i click connect
asterisk result is

notice 2339 chan_iax2.c8083 socket_process: rejected connection attempt form
10.0.1.176, request "gui at radio-gui" does not exits

could you help me?


-- 
U Can
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From hwstar at rodgers.sdcoxmail.com  Mon Feb 22 19:20:58 2010
From: hwstar at rodgers.sdcoxmail.com (hwstar at rodgers.sdcoxmail.com)
Date: Mon, 22 Feb 2010 11:20:58 -0800
Subject: [App_rpt-users] Test version of ACID 0.11
Message-ID: <20100222192058.YHLV9407.dukecmfep05.coxmail.com@dukecmimpo01.coxmail.com>


A test version of ACID version 0.11 is now available.

Go to:

http://dltest.allstarlink.org/

Download the ISO and burn a CD

Use the ISO to install the new version. 

Notes:

This test version has all source file fixes in SVN up to 2/20/2010. It does not resolve current issues open in Mantis as of 2/20/2010.


For testing only, do not use on production servers. The purpose of this release is to find problems before an official release is made. If you can take the time and volunteer to test it, we will get a better production release. 

The ISO gets its files from dltest.allstarlink.org so it
needs to be labelled as such. Be sure to switch to the standard ISO image available from dl.allstarlink.org when
0.11 is released.

Steve
WA6ZFT






From ted.freitas at mac.com  Wed Feb 24 00:33:01 2010
From: ted.freitas at mac.com (Ted G. Freitas)
Date: Tue, 23 Feb 2010 16:33:01 -0800
Subject: [App_rpt-users] Modulation Issue
Message-ID: <4B8473BD.9050901@mac.com>

I wanted to throw this question out there and see if anyone could help.

I am trying to use the XIPPR distribution of Centos from XELATEC, LLC.
that appears to have some neat features. However when we connect a URI
from DMK Engineering we are only getting about 1/2 volt peak to peak and
.7 volts peek to peak if we run the audio through the amplifier.

As a test we download the ACID distribution from app-rpt.qrvc.com and we
get about 1.7 volts peak to peak direct and about 3.2 volts peak to peak
through the amplifier.

Anyone know where I would start to look at why just using a different
Linux distribution would break this? Also if it turns out to be the
USBRADIO.c driver, how would I go about recompiling an updated version?

Thanks,
Ted/KE6YJC





From sales at qrvc.com  Wed Feb 24 03:10:03 2010
From: sales at qrvc.com (Stephen Rodgers)
Date: Tue, 23 Feb 2010 19:10:03 -0800
Subject: [App_rpt-users] Modulation Issue
In-Reply-To: <4B8473BD.9050901@mac.com>
References: <4B8473BD.9050901@mac.com>
Message-ID: <4B84988B.509@qrvc.com>

Ted G. Freitas wrote:
> I wanted to throw this question out there and see if anyone could help.
> 
> I am trying to use the XIPPR distribution of Centos from XELATEC, LLC.
> that appears to have some neat features. However when we connect a URI
> from DMK Engineering we are only getting about 1/2 volt peak to peak and
> .7 volts peek to peak if we run the audio through the amplifier.
> 
> As a test we download the ACID distribution from app-rpt.qrvc.com and we
> get about 1.7 volts peak to peak direct and about 3.2 volts peak to peak
> through the amplifier.
> 
> Anyone know where I would start to look at why just using a different
> Linux distribution would break this? Also if it turns out to be the
> USBRADIO.c driver, how would I go about recompiling an updated version?
> 
> Thanks,
> Ted/KE6YJC
> 
> 
> 
> _______________________________________________
> App_rpt-users mailing list
> App_rpt-users at qrvc.com
> http://qrvc.com/mailman/listinfo/app_rpt-users
> 

We don't spend time tracking what changes Steve Henke has made to XIPPR,  but my guess is that the files in the
asterisk/channels/xpmr directory are very likely to be significantly different between the two distributions.
XIPPR and ACID are forks and changes are not "cross pollinated" between them. We don't have the time nor the resources
to devote to that.

Steve
WA6ZFT




From nessenj at jimsoffice.org  Wed Feb 24 07:58:53 2010
From: nessenj at jimsoffice.org (James Nessen)
Date: Tue, 23 Feb 2010 23:58:53 -0800
Subject: [App_rpt-users] 2 DMK USB fobs for sale
In-Reply-To: <C60E5FFD-83F1-4246-A98E-3FA1C83C7C18@jimsoffice.org>
References: <C60E5FFD-83F1-4246-A98E-3FA1C83C7C18@jimsoffice.org>
Message-ID: <91a3416d1002232358r6a7e4ddfwec389ab3d4809c49@mail.gmail.com>

I found 2 more if anyone is still interested, again, please contact me
offlist.

Jim

On Thu, Nov 12, 2009 at 5:36 PM, James Nessen <nessenj at jimsoffice.org>wrote:

> Hi everyone,
>
> I have 2 DMK usb fobs that I am selling, if anyone is interested, please
> contact me off list.
>
> Thanks!
>
> Jim, K6JWN
>
>
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From nessenj at jimsoffice.org  Wed Feb 24 19:59:33 2010
From: nessenj at jimsoffice.org (James Nessen)
Date: Wed, 24 Feb 2010 11:59:33 -0800
Subject: [App_rpt-users] 2 DMK USB fobs for sale
In-Reply-To: <91a3416d1002232358r6a7e4ddfwec389ab3d4809c49@mail.gmail.com>
References: <C60E5FFD-83F1-4246-A98E-3FA1C83C7C18@jimsoffice.org>
	<91a3416d1002232358r6a7e4ddfwec389ab3d4809c49@mail.gmail.com>
Message-ID: <91a3416d1002241159o549304b7l59655383e0732c38@mail.gmail.com>

These are spoken for, thanks!

jim

On Tue, Feb 23, 2010 at 11:58 PM, James Nessen <nessenj at jimsoffice.org>wrote:

> I found 2 more if anyone is still interested, again, please contact me
> offlist.
>
> Jim
>
>
> On Thu, Nov 12, 2009 at 5:36 PM, James Nessen <nessenj at jimsoffice.org>wrote:
>
>> Hi everyone,
>>
>> I have 2 DMK usb fobs that I am selling, if anyone is interested, please
>> contact me off list.
>>
>> Thanks!
>>
>> Jim, K6JWN
>>
>>
>
>


-- 
Jim Nessen K6JWN
Email: nessenj at jimsoffice.org | Ph: 530.564.0039
IRLP: Nodes 3598,7358 | Echolink: 83598 K6JWN-R
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From ke2n at cs.com  Thu Feb 25 13:59:54 2010
From: ke2n at cs.com (Ken)
Date: Thu, 25 Feb 2010 08:59:54 -0500
Subject: [App_rpt-users] DTMF falsing
Message-ID: <8CC8439F0DD217F-7DFC-746A@webmail-m082.sysops.aol.com>

Is there a parameter that can be tweeked to reduce susceptibility of Asterisk to decoding voice as tones?? We have only one user who is bad at this, so it may not be fixable at the repeater end, but I think it would be interesting to try.? The parameter?would be the one that determines how long a tone pair must persist before it is considered valid.? I suppose this would be in the core Asterisk part of the program.
?
I understand that there is a trade off with "real" tones going out over the air. This would probably be more of an issue with a system that has autopatch....
?
Ken Jamrogowicz

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From kb4fxc at inttek.net  Thu Feb 25 15:20:15 2010
From: kb4fxc at inttek.net (David McGough)
Date: Thu, 25 Feb 2010 10:20:15 -0500 (EST)
Subject: [App_rpt-users] DTMF falsing
In-Reply-To: <8CC8439F0DD217F-7DFC-746A@webmail-m082.sysops.aol.com>
Message-ID: <Pine.LNX.4.44.1002251006340.362-100000@goliath.inttek.net>


Hi Ken,

I added some code that wrapped a simple finite-state-machine around the
existing tone detection code in chan_usbradio.c some time back. With this
code, DTMF tone passed thru the repeater until the FSM "accept" state was
reached--after a couple taps, as I recall. This worked well....In the
meantime, I discovered that if I turned OFF the RADIO_RELAX compiler flag
(from "make menuconfig" in the asterisk directory), I got even better
results. And, I subsequently removed the FSM code...

With RADIO_RELAX unset, I've had a few cases where DTMF from weak/noisey 
signals wouldn't decode. But, this hasn't been bad enough for me to 
re-enable RADIO_RELAX.

...Hope this helps!

73, David kb4fxc


On Thu, 25 Feb 2010, Ken wrote:

> Is there a parameter that can be tweeked to reduce susceptibility of Asterisk to decoding voice as tones?? We have only one user who is bad at this, so it may not be fixable at the repeater end, but I think it would be interesting to try.? The parameter?would be the one that determines how long a tone pair must persist before it is considered valid.? I suppose this would be in the core Asterisk part of the program.
> ?
> I understand that there is a trade off with "real" tones going out over the air. This would probably be more of an issue with a system that has autopatch....
> ?
> Ken Jamrogowicz
> 
>