[asterisk-dev] running pjsip testsuite

[Matthew Jordan]

On Thu, Apr 2, 2015 at 3:25 AM, Yaron Nachum <nachum.yaron at gmail.com> wrote:
> Hi everyone,
> I ran the test manually. Just setup a single endpoint  and using AMI I
> originanted a call to an extension which dials to another extension and send
> DTMF sequence using SendDTMF application.
>
> When I setup the endpoint with rfc4733 the dtmf is identified, but when I
> setup the endpoint with inband it is not identified. Using rtp debug I see
> that the rtp is sent and received.
>
> I did the same scenario with regular sip channel and the same happened.
>
> If anyone has a clue please get back to me.
>
> I will try to make the test with sipp.
>

Hey Yaron -

Can you attach a DEBUG log snippet from the Asterisk instance/channel
sending DTMF inband? In particular, the part where it does the
negotiation, along with sending the DTMF digit.

Thanks!

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org