The following issue arrose out of the bug report at <a href="http://bugs.digium.com/view.php?id=11489">http://bugs.digium.com/view.php?id=11489</a>.<br><br>I was asked by Eliel to post my thoughts to the developer list:<br>
<br>Beginning with asterisk-1.4.15 a bug concerning rfc2833 oob dtmf signalling on sip trunks was closed.<br>Unfortunately this fix opened a new issue:<br><br>Cisco carrier components (bt) use x-nse in the sdp message rather than the
<br>telephone-event (as defined in rfc2833) - although x-nse was meant to be a complement.<br>Concerning the interoperability between asterisk and those components this means<br>that rtp out-of-band signalling does not work anymore.
<br><br>I wrote a patch (for chan_sip) that, when dtmfmode=rfc2833 was set in the sip.conf, asterisk would use<br>rfc2833 even if the other side was not signalling this capability (resp. only signalled x-nse).<br><br>Eliel didn't feel comfortable about this as it is more of a work-around than a propper fix.
<br><br>My question is:<br>Which way of making the two systems interoperable at this level would<br>be fine to digium/the developer community (and would be most likely to be<br>accepted into main stream code base):<br><br>
1) modifying the code so that the existance of 'x-nse' in the sdp body is accepted as if<br>'telephone-event' was there<br>2) adding a config switch to sip.conf that would allow the admin to force using rfc2833 on
<br>a sip trunk even if the other side doesn't signal rfc2833 capability<br><br>Thanks for you comments (and if I overlooked the fact that someone has already<br>solved this, please let me know).<br><br>-Andreas<br><br>
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