>> Hopefully the initial poster still has the configuration to >> produce the files for you. >> >> Are you saying the DTMF logs I attached do not provide enough >> evidence to support the theory of the DTMF length being the >> cause of this issue? >> >> -Vladimir >> > Vladimir, > What was the Softphone/Version you were using to get this to fail. > > I'm using an old version of X-Lite, V3.0 build 56125 and with > asterisk 1.8.16.0 when in voicemail I was unable to get any errors. DTMF log > below. > > [2012-09-15 22:36:39.974909] DTMF[1706] channel.c: DTMF begin '1' received > on SIP/alec-00000009 > [2012-09-15 22:36:39.974985] DTMF[1706] channel.c: DTMF begin ignored '1' on > SIP/alec-00000009 > [2012-09-15 22:36:40.514978] DTMF[1706] channel.c: DTMF end '1' received on > SIP/alec-00000009, duration 560 ms > [2012-09-15 22:36:40.515037] DTMF[1706] channel.c: DTMF end passthrough '1' > on SIP/alec-00000009 > [2012-09-15 22:36:41.014955] DTMF[1706] channel.c: DTMF begin '2' received > on SIP/alec-00000009 > [2012-09-15 22:36:41.015009] DTMF[1706] channel.c: DTMF begin ignored '2' on > SIP/alec-00000009 > [2012-09-15 22:36:41.459045] DTMF[1706] channel.c: DTMF end '2' received on > SIP/alec-00000009, duration 460 ms > [2012-09-15 22:36:41.459089] DTMF[1706] channel.c: DTMF end passthrough '2' > on SIP/alec-00000009 > [2012-09-15 22:36:41.909042] DTMF[1706] channel.c: DTMF begin '3' received > on SIP/alec-00000009 > [2012-09-15 22:36:41.909093] DTMF[1706] channel.c: DTMF begin ignored '3' on > SIP/alec-00000009 > [2012-09-15 22:36:42.429177] DTMF[1706] channel.c: DTMF end '3' received on > SIP/alec-00000009, duration 540 ms > [2012-09-15 22:36:42.429236] DTMF[1706] channel.c: DTMF end passthrough '3' > on SIP/alec-00000009 > [2012-09-15 22:36:42.849091] DTMF[1706] channel.c: DTMF begin '4' received > on SIP/alec-00000009 > [2012-09-15 22:36:42.849185] DTMF[1706] channel.c: DTMF begin ignored '4' on > SIP/alec-00000009 > [2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end '4' received on > SIP/alec-00000009, duration 1660 > >
Alec, Please take a look at the case https://issues.asterisk.org/jira/browse/ASTERISK-20424?actionOrder=asc I uploaded the PCAP captured on the Soft Phone end and the RTP debug log. I ran the "old" Soft Phone", dialed *98, then entered "430", the application heard "444333000". Thank you, Vladimir -- _____________________________________________________________________ -- 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