, see the PR here:
https://github.com/pjsip/pjproject/pull/2328
And my patch here:
https://github.com/andreas-wehrmann/pjproject/commit/ef089cb53aa7570f7afda80a6a57f8b5778c86b4
"Advise your SIP provider" - haha, I had a good laugh...
All the bes
On 03/10/2019 16:24, Joshua C. Colp wrote:
In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately
codec negotiation is not written or implemented in the way you need. There are
some hints provided internally for outgoing legs but the result is still
ultimately
On 03.10.19 15:08, Administrator TOOTAI wrote:
Before calling the gatreway add
same = n,set(SIP_CODEC=alaw)
[...]
Hey there,
that doesn't work as it seems to be implemented for chan_sip only;
I'm using chan_pjsip; sorry if I didn't explain myself properly.
Anyway, in my case that would
Hello people,
I've ran into two problem that I can't seem to be able to solve on my own.
Here's my scenario (running Asterisk 13.28.1):
In short: - Asterisk behaves unexpectedly (at least to me) when
negotiating between endpoints
that have a different but intersecting set of codecs
