Re: [asterisk-users] PJSIP Lockup
On Mon, Mar 2, 2020 at 4:24 PM Nick Olsen wrote: > Thanks for the info, Joshua. > > Does PJSIP handle database access the same way Chan_sip did? We had a > number of boxes running chan_sip referencing the same mysql server without > issue. > > We're going to attempt to get a backtrace on the next occurance. We're > also going to run a local copy of the database on the same physical > asterisk instance and have the system reference it. Just to "throw > everything at the wall". > It uses the same underlying API and layer. It can do more frequent database access though due to queries and because PJSIP is multithreaded. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP Lockup
Thanks for the info, Joshua. Does PJSIP handle database access the same way Chan_sip did? We had a number of boxes running chan_sip referencing the same mysql server without issue. We're going to attempt to get a backtrace on the next occurance. We're also going to run a local copy of the database on the same physical asterisk instance and have the system reference it. Just to "throw everything at the wall". *Nick Olsen* Network Engineer Office: 321-408-5000 x103 Mobile: 321-794-0763 On Mon, Mar 2, 2020 at 1:58 PM Joshua C. Colp wrote: > On Mon, Mar 2, 2020 at 2:52 PM Nick Olsen < > n...@floridavirtualsolutions.com> wrote: > >> Hello All, >> I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But >> recently upgraded to attempt to resolve this issue. Using bundled PJSIP. >> The PBX is using mysql realtime for most functions. The Mysql server is >> on the same lan as the asterisk box. >> >> As more users have been moved to this box. It's become unstable. >> Randomly, I'll start seeing "WARNING[12667] taskprocessor.c: The >> 'pjsip/distributor-0173' task processor queue reached 500 scheduled >> tasks." >> >> At that time, Running "pjsip show contacts" and "pjsip show endpoints" >> returns nothing. And the box stops responding to all SIP. >> >> The only way I've found thus far to resolve the issue is a "service >> asterisk restart". >> >> I can confirm at the time of the issue running "asterisk -x 'core show >> taskprocessors' | grep 'distributor'" does show many items pending across >> all queues. And the number just increases. Normally when all is fine. >> They're all at 0. >> >> Google-foo hasn't produced anything for me outside issues from 13.x that >> claim to be resolved. Since asterisk isn't fully crashing, I don't think I >> can get backtrace. Someone please correct me if I'm wrong. >> Any ideas? Tips >> ? >> > > The wiki[1] has instructions for getting a backtrace for a deadlock from a > running process. It can be used to isolate why things are blocked. > Generally, though, when realtime is involved I've found that it usually > ends up being the database or that interaction in some way. Any hiccup or > issue there can result in blocking in Asterisk. > > [1] > https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No CID between Asterisk using IAX trunk
Could the difference be that you need to use type=friend for CID to work? Using type=peer we can forgo auth since we are not using public infrastructure. My other trunks do not have allowcallerid=yes so I will add that and test it. Thanks. On 02/03/20 12:54, Doug Lytle wrote: My Asterisk 13 IAX2 trunk posted below: type=friend trunk=yes allowcallerid=yes disallow=all allow=alaw allow=ulaw allow=gsm host=my.super.duper.host username=my.super.duper.username secret=my.super.duper.secret context=sip qualify=500 qualifysmoothing=yes requirecalltoken=no trunk=yes jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=300 maxjitterinterps=100 resyncthreshold=1500 tos=ef cos=5 Doug -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP Lockup
On Mon, Mar 2, 2020 at 2:52 PM Nick Olsen wrote: > Hello All, > I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But > recently upgraded to attempt to resolve this issue. Using bundled PJSIP. > The PBX is using mysql realtime for most functions. The Mysql server is on > the same lan as the asterisk box. > > As more users have been moved to this box. It's become unstable. Randomly, > I'll start seeing "WARNING[12667] taskprocessor.c: The > 'pjsip/distributor-0173' task processor queue reached 500 scheduled > tasks." > > At that time, Running "pjsip show contacts" and "pjsip show endpoints" > returns nothing. And the box stops responding to all SIP. > > The only way I've found thus far to resolve the issue is a "service > asterisk restart". > > I can confirm at the time of the issue running "asterisk -x 'core show > taskprocessors' | grep 'distributor'" does show many items pending across > all queues. And the number just increases. Normally when all is fine. > They're all at 0. > > Google-foo hasn't produced anything for me outside issues from 13.x that > claim to be resolved. Since asterisk isn't fully crashing, I don't think I > can get backtrace. Someone please correct me if I'm wrong. > Any ideas? Tips > ? > The wiki[1] has instructions for getting a backtrace for a deadlock from a running process. It can be used to isolate why things are blocked. Generally, though, when realtime is involved I've found that it usually ends up being the database or that interaction in some way. Any hiccup or issue there can result in blocking in Asterisk. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No CID between Asterisk using IAX trunk
My Asterisk 13 IAX2 trunk posted below: type=friend trunk=yes allowcallerid=yes disallow=all allow=alaw allow=ulaw allow=gsm host=my.super.duper.host username=my.super.duper.username secret=my.super.duper.secret context=sip qualify=500 qualifysmoothing=yes requirecalltoken=no trunk=yes jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=300 maxjitterinterps=100 resyncthreshold=1500 tos=ef cos=5 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP Lockup
Hello All, I'm using Asterisk 16.8.0 on a Centos 7 box. Previously 16.5.0, But recently upgraded to attempt to resolve this issue. Using bundled PJSIP. The PBX is using mysql realtime for most functions. The Mysql server is on the same lan as the asterisk box. As more users have been moved to this box. It's become unstable. Randomly, I'll start seeing "WARNING[12667] taskprocessor.c: The 'pjsip/distributor-0173' task processor queue reached 500 scheduled tasks." At that time, Running "pjsip show contacts" and "pjsip show endpoints" returns nothing. And the box stops responding to all SIP. The only way I've found thus far to resolve the issue is a "service asterisk restart". I can confirm at the time of the issue running "asterisk -x 'core show taskprocessors' | grep 'distributor'" does show many items pending across all queues. And the number just increases. Normally when all is fine. They're all at 0. Google-foo hasn't produced anything for me outside issues from 13.x that claim to be resolved. Since asterisk isn't fully crashing, I don't think I can get backtrace. Someone please correct me if I'm wrong. Any ideas? Tips? *Nick Olsen* Network Engineer Office: 321-408-5000 x103 Mobile: 321-794-0763 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No CID between Asterisk using IAX trunk
Not these particular two servers. On 02/03/20 12:16, Doug Lytle wrote: I am trying to troubleshoot two Asterisk servers that have an IAX2 trunk between them. Carlos, Had caller-id ever worked between these two systems? Doug -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No CID between Asterisk using IAX trunk
>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>> trunk between them. Carlos, Had caller-id ever worked between these two systems? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No CID between Asterisk using IAX trunk
I am trying to troubleshoot two Asterisk servers that have an IAX2
trunk between them. Calls come and go but there is no CallerID from the
remote server either way. One of the servers is running Asterisk 16 and
the other is an older 1.8 install (I know, I am trying to get permission
to update). The trunk between servers is very simple. Something like:
Server 1 (Mexico)
[panama]
type=peer
context=oficina
trunk=yes
disallow=all
allow=g729
qualify=yes
requirecalltoken=no
host=10.X.X.141
language=es
callerid=asreceived
Server 2 (Panama)
[mexico]
type=peer
context=oficina
trunk=yes
disallow=all
allow=g729
qualify=yes
requirecalltoken=no
host=10.Y.Y.5
language=es
callerid=asreceived
So from Panama to Mexico we use:
exten => _1XXX,1,Dial(IAX2/mexico/${EXTEN})
Call comes in and is answered but there is no CID in CDR or in the phone
display. Other trunks to other servers have no problem sending CID from
one server to the other (all using IAX). Any pointers?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
