<p>I have had that happen to me. Sometimes its a configuration change. Sometimes the database gets corrupted. Compare the file size of /var/lib/asterisk/astdb to your original if you have a copy.</p>
<p>The following may help.</p>
<p>mv /var/lib/asterisk/astdb /var/lib/asterisk/astdb.old</p>
<p>And restart the computer. Asterisk will rebuild the database on startup.<br></p>
<p>KB3ORS-Brian</p>
<div class="gmail_quote">On May 21, 2014 12:00 PM, <<a href="mailto:app_rpt-users-request@ohnosec.org">app_rpt-users-request@ohnosec.org</a>> wrote:<br type="attribution"><blockquote class="quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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Today's Topics:<br>
<br>
1. Status 139, signal 11 (Robert Newberry)<br>
2. Re: Status 139, signal 11 (Robert Newberry)<br>
3. Re: Connect SIP Phone to allstarlink box (Steve Wright)<br>
4. Re: Connect SIP Phone to allstarlink box (Roger Bly)<br>
5. Re: Connect SIP Phone to allstarlink box (<a href="mailto:mike@midnighteng.com">mike@midnighteng.com</a>)<br>
6. Re: Connect SIP Phone to allstarlink box (<a href="mailto:torben@klimt-online.com">torben@klimt-online.com</a>)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Tue, 20 May 2014 12:32:46 -0400<br>
From: Robert Newberry <<a href="mailto:N1XBM@amsat.org">N1XBM@amsat.org</a>><br>
To: <a href="mailto:app_rpt-users@ohnosec.org">app_rpt-users@ohnosec.org</a><br>
Subject: [App_rpt-users] Status 139, signal 11<br>
Message-ID:<br>
<CAM09xEai=3OvkfLZz+CCn2HWGKqymjAk21WKMFwz6AD4p=<a href="mailto:rEvQ@mail.gmail.com">rEvQ@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
I haven't worked on my repeater project in a while. Last time I ran my<br>
linux box a few months ago everything was fine. I turn it on today and I'm<br>
getting a repeating screen of status 139, signal 11.<br>
<br>
I tried attaching a picture of my screen but no joy on sending pics.<br>
<br>
I'm using cent os 5.9 on a he compaq Celeron D.<br>
<br>
Any ideas how to fix this? I hope I've provided enough information. If not<br>
I'll do my best to get more.<br>
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<br>
Message: 2<br>
Date: Tue, 20 May 2014 12:40:44 -0400<br>
From: Robert Newberry <<a href="mailto:N1XBM@amsat.org">N1XBM@amsat.org</a>><br>
To: <a href="mailto:app_rpt-users@ohnosec.org">app_rpt-users@ohnosec.org</a><br>
Subject: Re: [App_rpt-users] Status 139, signal 11<br>
Message-ID:<br>
<<a href="mailto:CAM09xEYQ3ZyZ80h9-YsY_XXTwE6RiKPt95Rfhz543NUenXmtrA@mail.gmail.com">CAM09xEYQ3ZyZ80h9-YsY_XXTwE6RiKPt95Rfhz543NUenXmtrA@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
I did find one thing searching the ohnosec archive. Back in 2012 someone<br>
had the same issue with setting confmode to yes. Setting it back to no<br>
fixed the issue.<br>
<br>
My concern is as far as I know that wasn't touched last time the box was<br>
running. Also how do I stop the continuous screen 9f scrolling errors.<br>
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Message: 3<br>
Date: Wed, 21 May 2014 09:06:52 +1200<br>
From: Steve Wright <<a href="mailto:stevewrightnz@gmail.com">stevewrightnz@gmail.com</a>><br>
To: <a href="mailto:App_rpt-users@ohnosec.org">App_rpt-users@ohnosec.org</a><br>
Subject: Re: [App_rpt-users] Connect SIP Phone to allstarlink box<br>
Message-ID:<br>
<<a href="mailto:CABSSkBU_iDeShPbJ4Z_92SoZJRnzHvLwUXZL0X8YBOb9rj46Xw@mail.gmail.com">CABSSkBU_iDeShPbJ4Z_92SoZJRnzHvLwUXZL0X8YBOb9rj46Xw@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
You have quite a few options when connecting SIP phones or other toys to<br>
the repeater network. You might add SIP config to the repeater node, and<br>
have SIP devices connect to it directly, or add another SIP server nearby<br>
and trunk the two machines.<br>
<br>
I looked at doing the same thing some ago, and I did manage to get SIP<br>
devices connecting directly to the repeater node, but it was just an<br>
experiment and was never put into service - but I could certainly dial in<br>
and connect audio to the node.<br>
<br>
I think if I was doing the job again, I would experiment with a separate<br>
machine for the SIP phone server software - perhaps install one of the<br>
commonly available Asterisk distributions for that task, and then peer that<br>
machine with the repeater node. This is added machinery, but I think it<br>
will make both of the systems much easier to build and maintain.<br>
<br>
Steve ZL1BHD<br>
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Message: 4<br>
Date: Tue, 20 May 2014 14:47:38 -0700<br>
From: Roger Bly <<a href="mailto:roger@rogerbly.com">roger@rogerbly.com</a>><br>
To: <a href="mailto:App_rpt-users@ohnosec.org">App_rpt-users@ohnosec.org</a><br>
Subject: Re: [App_rpt-users] Connect SIP Phone to allstarlink box<br>
Message-ID: <<a href="mailto:BF96F753-362F-46CD-BB10-4CDA0BD89D27@rogerbly.com">BF96F753-362F-46CD-BB10-4CDA0BD89D27@rogerbly.com</a>><br>
Content-Type: text/plain; charset=us-ascii<br>
<br>
Thanks. Just started looking into this also. Any tips on a way to secure SIP dial-in with unique password for each user?<br>
<br>
Regards,<br>
Roger Bly<br>
k6mwt<br>
<br>
<br>
On May 20, 2014, at 2:06 PM, Steve Wright <<a href="mailto:stevewrightnz@gmail.com">stevewrightnz@gmail.com</a>> wrote:<br>
<br>
> 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.<br>
><br>
> 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.<br>
><br>
> 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.<br>
><br>
> Steve ZL1BHD<br>
><br>
> _______________________________________________<br>
> App_rpt-users mailing list<br>
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------------------------------<br>
<br>
Message: 5<br>
Date: Tue, 20 May 2014 16:58:59 -0700<br>
From: <<a href="mailto:mike@midnighteng.com">mike@midnighteng.com</a>><br>
To: <a href="mailto:App_rpt-users@ohnosec.org">App_rpt-users@ohnosec.org</a><br>
Subject: Re: [App_rpt-users] Connect SIP Phone to allstarlink box<br>
Message-ID:<br>
<<a href="mailto:20140520165859.71befee5dbd13c5325dd1a521b4e73ee.79b403f6c8.wbe@email06.secureserver.net">20140520165859.71befee5dbd13c5325dd1a521b4e73ee.79b403f6c8.wbe@email06.secureserver.net</a>><br>
<br>
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<br>
Message: 6<br>
Date: Wed, 21 May 2014 08:55:38 +0200<br>
From: "<a href="mailto:torben@klimt-online.com">torben@klimt-online.com</a>" <<a href="mailto:torben@klimt-online.com">torben@klimt-online.com</a>><br>
To: "<a href="mailto:app_rpt-users@ohnosec.org">app_rpt-users@ohnosec.org</a>" <<a href="mailto:app_rpt-users@ohnosec.org">app_rpt-users@ohnosec.org</a>><br>
Subject: Re: [App_rpt-users] Connect SIP Phone to allstarlink box<br>
Message-ID: <CF45F236-47BD-4ACD-AD30-8194C7864DF0@mimectl><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
hi all,<br>
sorry for the late answer but i sleep when you write,Hi......<br>
@yes mike, it works fine<br>
@robert, i try to explain with my words what i do for working - if that is the right way i don`t know ;-)<br>
<br>
my english is not the best but i hope you can understand what i write<br>
<br>
<br>
<br>
greetings from bavaria tom dh6mbt<br>
<br>
1. Setup SIP-Phones in the sip.conf like mike write<br>
<br>
[6010]<br>
deny=<a href="http://0.0.0.0/0.0.0.0" target="_blank">0.0.0.0/0.0.0.0</a><br>
username=6010<br>
secret=6010<br>
dtmfmode=rfc2833<br>
canreinvite=no<br>
context=radio-control<br>
host=dynamic<br>
trustrpid=yes<br>
sendrpid=no<br>
type=friend<br>
nat=no<br>
port=5060<br>
qualify=yes<br>
qualifyfreq=60<br>
transport=udp<br>
encryption=no<br>
callgroup=<br>
pickupgroup=<br>
dial=SIP/6010<br>
mailbox=6010@device<mailto:<a href="mailto:mailbox">mailbox</a>=6010@device><br>
permit=<a href="http://0.0.0.0/0.0.0.0" target="_blank">0.0.0.0/0.0.0.0</a><br>
callerid=DH6MBT <6010><br>
<br>
do it also with [6020]<br>
<br>
2. edit extension.conf for dial in the repeater my repeater has the nodenr 28175<br>
<br>
[radio-control]<br>
exten => 28175,1,Answer<br>
exten => 28175,n,Wait(2)<br>
exten => 28175,n,Playback(rpt/node)<br>
exten => 28175,n,Playback(digits/2)<br>
exten => 28175,n,Playback(digits/8)<br>
exten => 28175,n,Playback(digits/1)<br>
exten => 28175,n,Playback(digits/7)<br>
exten => 28175,n,Playback(digits/5)<br>
exten => 28175,n,rpt,28175|Pv<br>
<br>
Pv stand for VOX by the phone, no other DTMF dials are needed<br>
if you take P instead of Pv you must dial *99 for talking and # to stop PTT<br>
<br>
3. edit the extension.conf to call from each SIP-phone to another<br>
<br>
exten => 6010,1,Dial(SIP/6010)<br>
exten => 6020,1,Dial(SIP/6020)<br>
<br>
3. edit the extension.conf to call from the repeater to the SIP-phone<br>
<br>
[autopatch]<br>
exten => 6010,1,Dial(SIP/6010)<br>
exten => 6020,1,Dial(SIP/6020)<br>
<br>
now you can talk from SIp to SIP by callig 6010 or 6020<br>
you can call from Sip the the repeater by dialing 28175<br>
and you can call your SIP from teh repeater by dialing *66010 or *66020 - hangup by *0<br>
<br>
<br>
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End of App_rpt-users Digest, Vol 63, Issue 38<br>
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</blockquote></div>