On Fri, Mar 7, 2014 at 12:07 PM, Daniel Pocock <[email protected]> wrote: > > > > Hi, > > I've recently observed an Asterisk 11.5 setup sending rfc2833 DTMF with > an SSRC that is distinct from the audio stream SSRC. > > It was discussed in this bug which is marked "FIXED", but it is still > happening, maybe it is a regression now: > > https://issues.asterisk.org/jira/browse/ASTERISK-16851
There isn't enough information here for someone to know whether or not this is the same issue. > Has this changed again because of the PjSIP changes? Nope. Both chan_pjsip and chan_sip use the same media stack for RFC 2833 DTMF generation. > Is the constantssrc option to be supported going forward? I'm not sure what option you are referring to. There isn't a constantssrc option in rtp.conf or in res_rtp_asterisk. > In case it is relevant: > - I'm using SIP > - the DTMF comes in to Asterisk using SIP INFO over TCP I'm assuming you have a DTMF generated by a SIP INFO request, which creates an RFC 2833 DTMF on an outbound channel. If that is the case, there are a number of open issues regarding SIP INFO to RFC 2833 conversion, but most of those have to do with the DTMF duration. You're welcome to open an issue for this, but I'd suggest gathering all of the standard logs illustrating the problem before doing so. Matt -- 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 -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
