[asterisk-users] DTMF behavior in asterisk 12 with PJSIP

[Matthew Jordan]

On Sun, Oct 26, 2014 at 3:22 AM, Yaron Nachum <nachum.yaron at gmail.com>
wrote:

> Hello all,
> We have recently upgraded some of our services to Asterisk 12 with PJSIP.
> We have 2 issues related to DTMF:
> 1. in the regular SIP channel we had DTMF auto mode, which adapted the
> DTMF settings according to the incoming INVITE - RFC2833 or inband. The is
> no such settings in PJSIP. Do you know is there is a plan to develop it?
>

No one that I'm aware of is currently working on that.

As Asterisk is an open source project, if having the 'auto' feature added
to the PJSIP stack is something you're interested in, you should consider
writing a patch for the project [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process


> 2. When we setup 2 peers, one RFC4733 and the other inband, the asterisk
> does not transcode the DTMF signals, therefore DTMF is not working. It used
> to work on release 11. This is really bad. Do you know of a solution to
> this issue? Maybe some settings?
>
>
That actually is a bug. You are most likely ending up in a native packet to
packet bridge (or a native remote bridge), which does not decode the RTP
stream. Hence, the inband DTMF or RFC 2833 DTMF is not being decoded and is
being passed to the other side. Please do open an issue for that [2]. Make
sure you provide a full DEBUG log, as that will illustrate what is actually
occurring.

Note that you can work around that issue by adding a feature flag to
whatever application caused the bridging to occur.

[2] https://issues.asterisk.org/jira

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