Hi all,
I’m trying to send an specific audio format (pcm, mono, samplerate=32000) to some devices which accept SIP and that kind of audio.
So, I’d like to know:
a) How can I get that audio into the conversation? (How can Asterisk use a custom codification).
b) Which are the steps to add that format?
c) How can the SDP negotiation be made to this payload type? I woud need to use a dynamic payload type?
It’s quite strange, I know, but I must get this working… 
so, Thank you!
That’s a developer question. You need to use something like the asterisk-dev mailing list, if you can get it to work (I got some early posts through, but all my recent ones seem to be being accepted by lists.digium.com and then lost - my last email to Bryan Johns has produced no reply, either).
You will need to write a format module and, if you want to generate tones, or do anything else that requires asterisk to transcode, a codec module. I think there is also a bit map of codecs that will need extending. I believe there is a dummy format module, as an example.
hey David, thanks for replying!
How and where could I write that format and codec modules? Where can I find that dummy format example?
as you can see… I’m quite new to this… 
THANK YOU!
There may not be examples. The simpler codecs may be good enough examples themselves.
You really need to be using developer channels. This is an end user forum.
The codec module goes in the codecs directory, in the source tree, and the format one in the formats one. Reading existing ones will probably give a clue as to which include files need updating.
Alright, Thanks a lot!!
I’ll have a look asap!