Re: [Linphone-developers] Trouble with RFC 2833 / 4733 DTMF

[Dominic]

Hi Russell,

I posted the SDP for both a successful and a failing call. Were
you able to see them both?

To answer your question, yes, voice works fine in both cases.
Linphone and asterisk agree on PCMU/8000 (ulaw). It doesn't look
like linphone offers that codec as an option in the initial
INVITE even though it is enabled in the settings, but it's listed
(a=rtpmap:0 PCMU/8000) in the 200 OK. For some reason, it looks
like the formatting of my last message got messed up in the
email. It's a lot easier to read on the FreePBX forum post if you
want to check it out there:
https://community.freepbx.org/t/help-debugging-rfc4733-dtmf/48958

You said that Asterisk wasn't indicating support for rfc-2883
dtmf or speex. I can confirm that speex is disabled in my
Asterisk codec settings, but it should support rfc-2883 dtmf.
Isn't that what the "a=rtpmap:98 telephone-event/8000" part of
the 200 OK response means? I apologize, I'm still quite new to
SIP/SDP/RTP.

> On Thu, 3 May 2018 20:08:32 -0400 Russell Treleaven <address@hidden> wrote:
> 
> Asterisk is not indicating support for rfc-2883 dtmf 16000khz or speex.
> What does linphone think was agreed upon?
> 
> Does voice work in this specific case?

-- 
Thank you,
Dominic