[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
<[email protected]
<mailto:[email protected]>> 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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