> 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 -- _____________________________________________________________________ -- 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
