On May 21, 2014 12:00 PM, <app_rpt-users-request@ohnosec.org> wrote:

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Today's Topics:

   1. Status 139, signal 11 (Robert Newberry)
   2. Re: Status 139, signal 11 (Robert Newberry)
   3. Re: Connect SIP Phone to allstarlink box (Steve Wright)
   4. Re: Connect SIP Phone to allstarlink box (Roger Bly)
   5. Re: Connect SIP Phone to allstarlink box (mike@midnighteng.com)
   6. Re: Connect SIP Phone to allstarlink box (torben@klimt-online.com)


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Message: 1
Date: Tue, 20 May 2014 12:32:46 -0400
From: Robert Newberry <N1XBM@amsat.org>
To: app_rpt-users@ohnosec.org
Subject: [App_rpt-users] Status 139, signal 11
Message-ID:
        <CAM09xEai=3OvkfLZz+CCn2HWGKqymjAk21WKMFwz6AD4p=rEvQ@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

I haven't worked on my repeater project in a while. Last time I ran my
linux box a few months ago everything was fine. I turn it on today and I'm
getting a repeating screen of status 139, signal 11.

I tried attaching a picture of my screen but no joy on sending pics.

I'm using cent os 5.9 on a he compaq Celeron D.

Any ideas how to fix this? I hope I've provided enough information. If not
I'll do my best to get more.
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Message: 2
Date: Tue, 20 May 2014 12:40:44 -0400
From: Robert Newberry <N1XBM@amsat.org>
To: app_rpt-users@ohnosec.org
Subject: Re: [App_rpt-users] Status 139, signal 11
Message-ID:
        <CAM09xEYQ3ZyZ80h9-YsY_XXTwE6RiKPt95Rfhz543NUenXmtrA@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

I did find one thing  searching the ohnosec archive.  Back in 2012 someone
had the same issue with setting confmode to yes. Setting it back to no
fixed the issue.

My concern is as far as I know that wasn't touched last time the box was
running.  Also how do I stop the continuous screen 9f scrolling errors.
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Message: 3
Date: Wed, 21 May 2014 09:06:52 +1200
From: Steve Wright <stevewrightnz@gmail.com>
To: App_rpt-users@ohnosec.org
Subject: Re: [App_rpt-users] Connect SIP Phone to allstarlink box
Message-ID:
        <CABSSkBU_iDeShPbJ4Z_92SoZJRnzHvLwUXZL0X8YBOb9rj46Xw@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

You have quite a few options when connecting SIP phones or other toys to
the repeater network.  You might add SIP config to the repeater node, and
have SIP devices connect to it directly, or add another SIP server nearby
and trunk the two machines.

I looked at doing the same thing some ago, and I did manage to get SIP
devices connecting directly to the repeater node, but it was just an
experiment and was never put into service - but I could certainly dial in
and connect audio to the node.

I think if I was doing the job again, I would experiment with a separate
machine for the SIP phone server software - perhaps install one of the
commonly available Asterisk distributions for that task, and then peer that
machine with the repeater node.  This is added machinery, but I think it
will make both of the systems much easier to build and maintain.

Steve ZL1BHD
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Message: 4
Date: Tue, 20 May 2014 14:47:38 -0700
From: Roger Bly <roger@rogerbly.com>
To: App_rpt-users@ohnosec.org
Subject: Re: [App_rpt-users] Connect SIP Phone to allstarlink box
Message-ID: <BF96F753-362F-46CD-BB10-4CDA0BD89D27@rogerbly.com>
Content-Type: text/plain; charset=us-ascii

Thanks.   Just started looking into this also.   Any tips on a way to secure SIP dial-in with unique password for each user?

Regards,
Roger Bly
k6mwt


On May 20, 2014, at 2:06 PM, Steve Wright <stevewrightnz@gmail.com> wrote:

> You have quite a few options when connecting SIP phones or other toys to the repeater network.  You might add SIP config to the repeater node, and have SIP devices connect to it directly, or add another SIP server nearby and trunk the two machines.
>
> I looked at doing the same thing some ago, and I did manage to get SIP devices connecting directly to the repeater node, but it was just an experiment and was never put into service - but I could certainly dial in and connect audio to the node.
>
> I think if I was doing the job again, I would experiment with a separate machine for the SIP phone server software - perhaps install one of the commonly available Asterisk distributions for that task, and then peer that machine with the repeater node.  This is added machinery, but I think it will make both of the systems much easier to build and maintain.
>
> Steve ZL1BHD
>
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Message: 5
Date: Tue, 20 May 2014 16:58:59 -0700
From: <mike@midnighteng.com>
To: App_rpt-users@ohnosec.org
Subject: Re: [App_rpt-users] Connect SIP Phone to allstarlink box
Message-ID:
        <20140520165859.71befee5dbd13c5325dd1a521b4e73ee.79b403f6c8.wbe@email06.secureserver.net>

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Message: 6
Date: Wed, 21 May 2014 08:55:38 +0200
From: "torben@klimt-online.com" <torben@klimt-online.com>
To: "app_rpt-users@ohnosec.org" <app_rpt-users@ohnosec.org>
Subject: Re: [App_rpt-users] Connect SIP Phone to allstarlink box
Message-ID: <CF45F236-47BD-4ACD-AD30-8194C7864DF0@mimectl>
Content-Type: text/plain; charset="iso-8859-1"

hi all,
sorry for the late answer but i sleep when you write,Hi......
@yes mike, it works fine
@robert, i try to explain with my words what i do for working - if that is the right way i don`t know ;-)

my english is not the best but i hope you can understand what i write



greetings from bavaria tom dh6mbt

1. Setup SIP-Phones in the sip.conf like mike write

[6010]
deny=0.0.0.0/0.0.0.0
username=6010
secret=6010
dtmfmode=rfc2833
canreinvite=no
context=radio-control
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
encryption=no
callgroup=
pickupgroup=
dial=SIP/6010
mailbox=6010@device<mailto:mailbox=6010@device>
permit=0.0.0.0/0.0.0.0
callerid=DH6MBT <6010>

do it also with [6020]

2. edit extension.conf for dial in the repeater my repeater has the nodenr 28175

[radio-control]
exten => 28175,1,Answer
exten => 28175,n,Wait(2)
exten => 28175,n,Playback(rpt/node)
exten => 28175,n,Playback(digits/2)
exten => 28175,n,Playback(digits/8)
exten => 28175,n,Playback(digits/1)
exten => 28175,n,Playback(digits/7)
exten => 28175,n,Playback(digits/5)
exten => 28175,n,rpt,28175|Pv

Pv stand for VOX by the phone, no other DTMF dials are needed
if you take P instead of Pv you must dial *99 for talking and # to stop PTT

3. edit the extension.conf to call from each SIP-phone to another

exten => 6010,1,Dial(SIP/6010)
exten => 6020,1,Dial(SIP/6020)

3. edit the extension.conf to call from the repeater to the SIP-phone

[autopatch]
exten => 6010,1,Dial(SIP/6010)
exten => 6020,1,Dial(SIP/6020)

now you can talk from SIp to SIP by callig 6010 or 6020
you can call from Sip the the repeater by dialing 28175
and  you can call your SIP from teh repeater by dialing *66010 or *66020 - hangup by *0


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