[asterisk-users] Queue don't call Interface PJSIP

Joshua C. Colp jcolp at sangoma.com
Tue Aug 18 07:07:50 CDT 2020


On Tue, Aug 18, 2020 at 9:00 AM Roberto <
roberto.medola at gasparimsantos.com.br> wrote:

> Hi Joshua, thanks for answer.
> In this particular test my extension is on a simple network. There is no
> NAT, just an asterisk running on a virtual machine on a 7- 64bit CentOs. I
> am simulating an environment to be able to use PJSIP on my client. And even
> in this small environment, my extension does not call.
>
> My problem with NAT was with SIP "one way audio" on a client. All of this
> testing is to replace SIP with PJSIP on this client. But as the queue is
> unable to call a PJSIP extension, the migration project on the client is
> stopped.
>
>
> I tried to separate the debug file, but it seems to me that in asterisk
> 17.16.0, there is a problem or I did not know how to configure it, because
> the log did not generate it either.
> on console:
> "pjsip set logger on"
> "pjsip set history on"
>
> on file Logger.conf:
> debbuger => debug, trace
>
> asterisk -rx "reload"
>
> Make same calls, and opening the file only the following appears:
>
> [2020-08-18 08:46:47.778] Asterisk 17.6.0 built by root @ asterisk-homolog
> on a x86_64 running Linux on 2020-08-13 22:40:11 UTC\
>

The PJSIP packet logging are verbose messages, if verbose is enabled on
console or file they will show up there. The history module also uses CLI
commands to examine the history log.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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