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

[Roberto]

[SOLVED]!!!

My function that changed the callerid was returning an invalid number. 
Although the asterisk sends the call, the SIP header was wrong and the 
extension did not ring

Thanks.


Em 18/08/2020 09:07, Joshua C. Colp escreveu:
> On Tue, Aug 18, 2020 at 9:00 AM Roberto 
> <roberto.medola at gasparimsantos.com.br 
> <mailto: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 <http://www.sangoma.com> and 
> www.asterisk.org <http://www.asterisk.org>
>

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