[asterisk-users] PJSIP signaling question

[George Joseph]

On Tue, Mar 1, 2016 at 5:37 PM, Kevin Long <kevin.long at haloprivacy.com>
wrote:

>
>
> Interesting, thanks George. I pulled Asterisk 13 from git and the new
> pjproject from the SVN and will test accordingly .
>

​Yeah, actually you do need Asterisk 13 from git because pjproject
deprecated an api in trunk and we only handle that in the current git 13
branch.​



>
>
> I have a few more questions about PJSIP in Asterisk 13:
>
>
> 1.  Is there any way to list current ongoing calls and see what codecs are
> being used in the RTP streams?  With chan_sip,  “sip show channels” did
> this.
>
> 2. Also with a PJSIP initiated call, is there a way to see how man RTP
> packets have been sent and received for the call , I am debugging some
> intermittent 1-way and no-way audio on calls , and I am having trouble
> figuring out fi it is the client, firewall, or Asterisk/pjsip that is the
> culprit .
>

​Unfortunately, no to both (at least that I'm aware of).   I remember
looking at the channel stats a long while back and for some reason didn't
go any further.  I can re-look.



>
>
> Regards,
>
> Kevin Long
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