[asterisk-users] asterisk pjsip webrtc rtp to private IP

[Joshua C. Colp]

On Thu, Dec 12, 2019 at 11:40 AM marek <cervajs64 at gmail.com> wrote:

> examples of "interesting" information like ICE result and howto make
> "minimal" configuration of pjproject.conf
>
> i.e.
>
> for  debugging app_queue.so
>
> core set debug 5 app_queue.so
>
> for debugging RTP
>
> core set debug 10 rtp_engine
> core set debug 10 res_rtp_asterisk
>
> rtp set debug on
>
> logger.conf
>
> rtp => debug,verbose(5)
>
>
> so i mean
>
> in
> https://github.com/asterisk/asterisk/blob/master/configs/samples/pjproject.conf.sample
>
> by few examples try to explain  what usefull info i can get
>
>
> set
>
> [startup]
> log_level=6
> type=startup
>
> and dig what's usefull is not very productive
>
> btw we are using tools like sipcapture.org,voipmonitor.org, callstats.io,
> elasticsearch+filebeat, ... but without informations whats happening inside
> asterisk is harder to solve problems
>
>
Please file an issue[1] with these specific examples and any others so it's
not lost.

[1] https://issues.asterisk.org/jira

-- 
Joshua C. Colp
Senior Software Developer
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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