[asterisk-users] DMTF Mode

[Dan Journo]

> I just added alaw to the provider's list, so now it reads (gsm, ulaw, alaw) and its started working in a fashion.

> The DTMF tones keep "getting stuck". I press a number on the sip phone, and the other party hears a tone. But every few tones, it gets stuck and they hear a long tone of about 3 seconds and then it goes off.

Here's the debug log for two DTMF tones. The first was fine. The second got stuck.

[2010-10-13 16:25:16] DEBUG[3287]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4)
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2129 ast_rtp_change_source: Changing ssrc from 775511001 to 1841818300 due to a source change
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2129 ast_rtp_change_source: Changing ssrc from 381691761 to 1746631866 due to a source change
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:940 ast_rtcp_read: Got RTCP report of 72 bytes
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4)
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:635 create_dtmf_frame: Sending dtmf: 53 (5), at 91.110.53.170
[2010-10-13 16:25:16] DEBUG[18139]: channel.c:4565 ast_generic_bridge: Got DTMF begin on channel (SIP/kesher_201-00000381)
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2117 ast_rtp_new_source: Setting the marker bit due to a source update
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2117 ast_rtp_new_source: Setting the marker bit due to a source update

[2010-10-13 16:25:16] DEBUG[18139]: channel.c:4882 ast_channel_bridge: Bridge stops bridging channels SIP/kesher_201-00000381 and SIP/magrathea-00000382
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2129 ast_rtp_change_source: Changing ssrc from 1841818300 to 455288846 due to a source change
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2129 ast_rtp_change_source: Changing ssrc from 1746631866 to 340402601 due to a source change
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4)
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4)
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4)
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:635 create_dtmf_frame: Sending dtmf: 53 (5), at 91.110.53.170
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:738 process_rfc2833: - RTP 2833 Event: 00000005 (len = 4)
[2010-10-13 16:25:16] DEBUG[18139]: channel.c:4565 ast_generic_bridge: Got DTMF end on channel (SIP/kesher_201-00000381)
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2117 ast_rtp_new_source: Setting the marker bit due to a source update
[2010-10-13 16:25:16] DEBUG[18139]: rtp.c:2117 ast_rtp_new_source: Setting the marker bit due to a source update