[asterisk-dev] running pjsip testsuite

[Yaron Nachum]

Hi Mathew,
It is attached.

On Thu, Apr 2, 2015 at 4:56 PM, Matthew Jordan <mjordan at digium.com> wrote:

> 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
>
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