On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: > > On Fri, 17 Apr 2009, Jeff LaCoursiere wrote: > >>> I went ahead and switched to SIP just for grins, and made sure >>> "dtmfmode=rfc2833" is in the peer config on both sides and in the entry >>> for the phone. So now it is: >>> >>> polycom501---SIP/ulaw--->ast1---SIP/g729--->ast2---IAX/ulaw--->ITSP > > A bit more information. ast1 is running 1.4.23.1 and I noticed a debug line > in rtp.c: > > if (rtpdebug || option_debug > 2) > ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", > event, len); > > So I set debug to 10 and caught this line: > > [Apr 17 17:28:02] DEBUG[27264] rtp.c: - RTP 2833 Event: 00000002 (len = 4) > > So I guess that proves that from the phone to ast1 RFC2833 is in effect (I > did actually press the digit '2', which I assume is the event code above?). > > I tried to do the same on ast2, which is running 1.4.22.1, and with debug set > to 10 I did *not* get this message, which makes me think that RCF2833 is NOT > in effect for the trunk between ast1 and ast2. Is that reasonable? >
The main problem turned out to be at my ITSP, and is now resolved. The question remains for me, though, how to interpret the debug lines I was able to catch (or not) above. How do you really know if RFC2833 signalling is being received? I caught the debug message on ast1 but not on ast2. I am using ulaw between ast2 and the ITSP, and I am now wondering if the DTMF is being sent inband on that last leg since I could not catch the debug messages on ast2. Perhaps what they did to fix on their end is simply remove compression between themselves and the PSTN. I would really like a concrete method of verifying that DTMF signalling is being sent out of band on my outbound IAX link. Any ideas? Thanks, j _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users