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

[Roberto]

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\

Em 17/08/2020 18:57, Joshua C. Colp escreveu:
> On Mon, Aug 17, 2020 at 6:16 PM Roberto 
> <roberto.medola at gasparimsantos.com.br 
> <mailto:roberto.medola at gasparimsantos.com.br>> wrote:
>
>     Hello.
>
>
>     I am having a lot of problems with SIP through NAT. So, I decided
>     to adopt PJSIP. However, I am not able to make the extensions ring
>     when receiving a call from the queue. I'm using telnet to include
>     the extension and on the asterisk console, it even shows Called
>     PJSIP/6001, but the extension doesn't ring. If I call from
>     extension to extension, it works normally.
>
>
> Can you describe the actual network setup further? Is the endpoint 
> behind NAT or merely Asterisk? I ask because there is no NAT 
> configuration for the endpoint, which if it is behind one can be 
> problematic. Failing that you'll need to provide a SIP trace using 
> "pjsip set logger on" to show the actual SIP traffic flowing (and 
> where to).
>
> -- 
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com <http://www.sangoma.com> and 
> www.asterisk.org <http://www.asterisk.org>
>

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