Just so you guys know, I upgraded to 1.4.5 (which was released yesterday after Russell and I fixed a bug in chan_sip). I do not have my DTMF problems thus far, so I would recommend trying 1.4.5 which had a lot of DTMF things fixed post-1.4.4.
Leif Madsen On 6/15/07, Stephan Monette <[EMAIL PROTECTED]> wrote:
This is a known issue with Asterisk since it's not 100% compliant with RFC2833. We have the same problem using 1.2.x or 1.4.4. This is something to do with Variable Length DTMF tones in RFC2833. Thanks. Stephan Monette Unlimitel Inc. Tel.: 1 (877) 464-6638, x221 Leif Madsen wrote: > On 6/13/07, Henry L.Coleman <[EMAIL PROTECTED]> wrote: >> 1. A three way conference call is set up by party "B" between "A" and >> "B"and "C" >> party "A" has to navigate party "C"'s IVR but the DTMF signaling doesn't >> work. >> Party "B" is a Grandstream GXP 2000 (SIP) with signalling set to <via >> RTP (RFC2833)> The trunks are IAX (unlimitel) > > Hey Henry > > Just to let you know, I'm having the exact same problem (DTMF with > RFC2833 on Asterisk, using the SIP channel). > > When I call into a conference and it asks for the PIN, it doesn't > accept it -- the strangest thing is that Asterisk "sees" the DTMF via > the logger.conf, console => dtmf settings. > > I'm still trying to track this down and will update this thread with a > bug number once I get some more information. > > For now, I'm piggy-backing onto this bug, but I think I gotta open > something separate (I don't think the issues are related anymore): > > http://bugs.digium.com/view.php?id=9959 > > More information to follow when I get it. >
-- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk
