Hello Mathew, Thank you for the reply. I will open an issue and send debug information.
Can you explain more about the workaround? A reference to the documentation would be fine. Thanks again, Yaron. On Sun, Oct 26, 2014 at 10:46 PM, Matthew Jordan <[email protected]> wrote: > > > On Sun, Oct 26, 2014 at 3:22 AM, Yaron Nachum <[email protected]> > 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
