[root@lib1 asterisk]# asterisk -vvvvvvdddddr Asterisk 1.4.0-beta3, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf Found == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf Found Connected to Asterisk 1.4.0-beta3 currently running on lib1 (pid = 20207) -- Remote UNIX connection Verbosity was 3 and is now 6 lib1*CLI> sip debug SIP Debugging enabled -- Attempting call on SIP/bart@cop.libretel.com for 813@record-request:1 (Retry 1) -- parse_srv: SRV mapped to host lib1.libretel.com, port 5060 Audio is at 69.90.155.71 port 13130 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 69.90.155.71:5060: INVITE sip:bart@cop.libretel.com SIP/2.0 Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK4d5ec8ad;rport From: "testcaller" ;tag=as24f9848b To: Contact: Call-ID: 1c6c6f345cdf2b4152e89db1244309fd@69.90.155.71 CSeq: 102 INVITE User-Agent: Vivox Server Max-Forwards: 70 Date: Tue, 24 Oct 2006 20:19:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 276 v=0 o=root 20207 20207 IN IP4 69.90.155.71 s=session c=IN IP4 69.90.155.71 t=0 0 m=audio 13130 RTP/AVP 0 8 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- lib1*CLI> <--- SIP read from 69.90.155.71:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK4d5ec8ad;rport=5028 From: "testcaller" ;tag=as24f9848b To: Call-ID: 1c6c6f345cdf2b4152e89db1244309fd@69.90.155.71 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- lib1*CLI> <--- SIP read from 69.90.155.71:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK4d5ec8ad;rport=5028 From: "testcaller" ;tag=as24f9848b To: ;tag=b4b01b74e1244d2c9a87767ff437efae Call-ID: 1c6c6f345cdf2b4152e89db1244309fd@69.90.155.71:5060 CSeq: 102 INVITE Record-Route: User-Agent: RTC/1.3 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- lib1*CLI> <--- SIP read from 69.90.155.71:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK4d5ec8ad;rport=5028 From: "testcaller" ;tag=as24f9848b To: ;tag=b4b01b74e1244d2c9a87767ff437efae Call-ID: 1c6c6f345cdf2b4152e89db1244309fd@69.90.155.71:5060 ------> ID includes port# CSeq: 102 INVITE Record-Route: Contact: User-Agent: RTC/1.3 Content-Type: application/sdp Content-Length: 255 v=0 o=- 0 0 IN IP4 68.166.120.178 s=session c=IN IP4 68.166.120.178 b=CT:1000 t=0 0 m=audio 23246 RTP/AVP 0 8 4 101 a=rtcp:20521 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (11 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 68.166.120.178:23246 Got unsupported a:rtcp in SDP offer Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xd (g723|ulaw |alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 68.166.120.178:23246 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 69.90.155.71, port 5060 Transmitting (no NAT) to 69.90.155.71:5060: ACK sip:68.166.120.178:15076 SIP/2.0 Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK2cf1d2f7;rport Route: From: "testcaller" ;tag=as24f9848b To: ;tag=b4b01b74e1244d2c9a87767ff437efae Contact: Call-ID: 1c6c6f345cdf2b4152e89db1244309fd@69.90.155.71 CSeq: 102 ACK User-Agent: Vivox Server Max-Forwards: 70 Content-Length: 0 --- > Channel SIP/cop.libretel.com-08b32230 was answered. -- Executing [813@record-request:1] Answer("SIP/cop.libretel.com-08b32230", "") in new stack -- Executing [813@record-request:2] Set("SIP/cop.libretel.com-08b32230", "TIMEOUT(absolute)=300") in new stack -- Channel will hangup at 2006-10-24 20:24:10 UTC. -- Executing [813@record-request:3] Playback("SIP/cop.libretel.com-08b32230", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'en') [Oct 24 16:19:10] WARNING[29349]: rtp.c:1073 ast_rtp_read: RTP Read too short lib1*CLI> <--- SIP read from 69.90.155.71:5060 ---> BYE sip:testcaller@69.90.155.71:5028 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 69.90.155.71;branch=z9hG4bKb3a9.27286cf6.0 Via: SIP/2.0/UDP 68.166.120.178:15076 Max-Forwards: 16 From: ;tag=b4b01b74e1244d2c9a87767ff437efae To: "testcaller" ;tag=as24f9848b Call-ID: 1c6c6f345cdf2b4152e89db1244309fd@10.1.1.188:4127 CSeq: 1 BYE User-Agent: RTC/1.3 Content-Length: 0 Route: <-------------> --- (12 headers 0 lines) --- lib1*CLI> <--- Transmitting (no NAT) to 69.90.155.71:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 69.90.155.71;branch=z9hG4bKb3a9.27286cf6.0;received=69.90.155.71 Via: SIP/2.0/UDP 68.166.120.178:15076 From: ;tag=b4b01b74e1244d2c9a87767ff437efae To: "testcaller" ;tag=as24f9848b Call-ID: 1c6c6f345cdf2b4152e89db1244309fd@10.1.1.188:4127 CSeq: 1 BYE User-Agent: Vivox Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> -- Executing [813@record-request:4] Echo("SIP/cop.libretel.com-08b32230", "") in new stack lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> Test with debugging turned on lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> lib1*CLI> logger reload == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted Asterisk Queue Logger restarted [Oct 24 16:19:41] DEBUG[20264]: pbx_spool.c:391 scan_service: Delaying retry since we're currently running '¸ča' -- Attempting call on SIP/bart@cop.libretel.com for 813@record-request:1 (Retry 1) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:14889 sip_request_call: Asked to create a SIP channel with formats: 0x40 (slin ) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4203 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) -- parse_srv: SRV mapped to host lib1.libretel.com, port 5060 [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:3699 sip_new: *** Our native formats are 0x4 (ulaw) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:3700 sip_new: *** Joint capabilities are 0x0 (nothing) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:3701 sip_new: *** Our capabilities are 0xd (g723|ulaw|alaw) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:3702 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:3704 sip_new: *** Our preferred formats from the incoming channel are 0x40 (sl in) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:3725 sip_new: This channel will not be able to handle video. [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:2774 sip_call: Outgoing Call for bart [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:2934 update_call_counter: Updating call counter for outgoing call [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:2782 sip_call: Our T38 capability (0), joint T38 capability (0) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:6007 add_sdp: ** Our capability: 0xd (g723|ulaw|alaw) Video flag: False [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:6008 add_sdp: ** Our prefcodec: 0x40 (slin) Audio is at 69.90.155.71 port 11980 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:6159 add_sdp: -- Done with adding codecs to SDP [Oct 24 16:19:46] DEBUG[29449]: channel.c:2226 ast_internal_timing_enabled: Internal timing is disabled (option_internal_ timing=0 chan->timingfd=-1) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:6204 add_sdp: Done building SDP. Settling with this capability: 0xd (g723|ulaw |alaw) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 0: INVITE sip:bart@cop.libretel.com SIP/2.0 (40) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 1: Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4b K1625db0f;rport (63) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 2: From: "testcaller" ;tag=as6bb15a36 (68) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 3: To: (31) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 4: Contact: (43) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 5: Call-ID: 4f385fa36cff57f332bf25175d2bd33e@69.90. 155.71 (54) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 6: CSeq: 102 INVITE (16) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 7: User-Agent: Vivox Server (24) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 8: Max-Forwards: 70 (16) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 9: Date: Tue, 24 Oct 2006 20:19:46 GMT (35) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER , SUBSCRIBE, NOTIFY (66) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 11: Supported: replaces (19) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 12: Content-Type: application/sdp (29) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 13: Content-Length: 276 (19) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4434 parse_request: Header 14: (0) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: v=0 (3) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: o=root 20207 20207 IN IP4 69.90.155.71 (38) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: s=session (9) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: c=IN IP4 69.90.155.71 (21) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: t=0 0 (5) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: m=audio 11980 RTP/AVP 0 8 4 101 (31) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: a=fmtp:101 0-16 (15) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: a=silenceSupp:off - - - - (25) [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:4466 parse_request: Line: a=sendrecv (10) Reliably Transmitting (no NAT) to 69.90.155.71:5060: INVITE sip:bart@cop.libretel.com SIP/2.0 Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK1625db0f;rport From: "testcaller" ;tag=as6bb15a36 To: Contact: Call-ID: 4f385fa36cff57f332bf25175d2bd33e@69.90.155.71 CSeq: 102 INVITE User-Agent: Vivox Server Max-Forwards: 70 Date: Tue, 24 Oct 2006 20:19:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 276 v=0 o=root 20207 20207 IN IP4 69.90.155.71 s=session c=IN IP4 69.90.155.71 t=0 0 m=audio 11980 RTP/AVP 0 8 4 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- [Oct 24 16:19:46] DEBUG[29449]: chan_sip.c:1927 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packe t: Id #9 lib1*CLI> <--- SIP read from 69.90.155.71:5060 ---> SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK1625db0f;rport=5028 From: "testcaller" ;tag=as6bb15a36 To: Call-ID: 4f385fa36cff57f332bf25175d2bd33e@69.90.155.71 CSeq: 102 INVITE Content-Length: 0 <-------------> [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 0: SIP/2.0 100 trying -- your call is important to us (50) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 1: Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4b K1625db0f;rport=5028 (68) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 2: From: "testcaller" ;tag=as6bb15a36 (68) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 3: To: (31) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 4: Call-ID: 4f385fa36cff57f332bf25175d2bd33e@69.90. 155.71 (54) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 5: CSeq: 102 INVITE (16) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 6: Content-Length: 0 (17) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 7: (0) --- (7 headers 0 lines) --- [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4254 find_call: = Found Their Call ID: '4f385fa36cff57f332bf25175d2bd33e@69.90 .155.71' Their Tag Our tag: as6bb15a36 Our ID: '4f385fa36cff57f332bf25175d2bd33e@69.90.155.71' [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:2069 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #9 - INVITE (got response) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:2078 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining pack et) on '4f385fa36cff57f332bf25175d2bd33e@69.90.155.71' Request 102: Found [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:11389 handle_response_invite: SIP response 100 to standard invite lib1*CLI> <--- SIP read from 69.90.155.71:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK1625db0f;rport=5028 From: "testcaller" ;tag=as6bb15a36 To: ;tag=7a11619fdc424bf2885ca1c154c560a1 Call-ID: 4f385fa36cff57f332bf25175d2bd33e@69.90.155.71:5060 CSeq: 102 INVITE Record-Route: User-Agent: RTC/1.3 Content-Length: 0 <-------------> [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 1: Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4b K1625db0f;rport=5028 (68) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 2: From: "testcaller" ;tag=as6bb15a36 (68) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 3: To: ;tag=7a11619fdc42 4bf2885ca1c154c560a1 (68) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 4: Call-ID: 4f385fa36cff57f332bf25175d2bd33e@69.90. 155.71:5060 (59) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 5: CSeq: 102 INVITE (16) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 6: Record-Route: (59) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 7: User-Agent: RTC/1.3 (19) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 8: Content-Length: 0 (17) [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 9: (0) --- (9 headers 0 lines) --- [Oct 24 16:19:46] DEBUG[20269]: chan_sip.c:4254 find_call: = Found Their Call ID: '4f385fa36cff57f332bf25175d2bd33e@69.90 .155.71' Their Tag Our tag: as6bb15a36 Our ID: '4f385fa36cff57f332bf25175d2bd33e@69.90.155.71:5060' [Oct 24 16:19:47] DEBUG[20269]: chan_sip.c:2078 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining pack et) on '4f385fa36cff57f332bf25175d2bd33e@69.90.155.71' Request 102: Found [Oct 24 16:19:47] DEBUG[20269]: chan_sip.c:11389 handle_response_invite: SIP response 180 to standard invite [Oct 24 16:19:47] DEBUG[20269]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/cop.libretel.com-08b38fb0 [Oct 24 16:19:47] DEBUG[20229]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - cop.libretel.com [Oct 24 16:19:47] DEBUG[20229]: chan_sip.c:14831 sip_devicestate: Checking device state for peer cop.libretel.com [Oct 24 16:19:47] DEBUG[20229]: devicestate.c:287 do_state_change: Changing state for SIP/cop.libretel.com - state 4 (Inv alid) [Oct 24 16:19:47] DEBUG[29450]: app_queue.c:546 changethread: Device 'SIP/cop.libretel.com' changed to state '4' (Invalid ) but we don't care because they're not a member of any queue. lib1*CLI> <--- SIP read from 69.90.155.71:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK1625db0f;rport=5028 From: "testcaller" ;tag=as6bb15a36 To: ;tag=7a11619fdc424bf2885ca1c154c560a1 Call-ID: 4f385fa36cff57f332bf25175d2bd33e@69.90.155.71:5060 CSeq: 102 INVITE Record-Route: Contact: User-Agent: RTC/1.3 Content-Type: application/sdp Content-Length: 255 v=0 o=- 0 0 IN IP4 68.166.120.178 s=session c=IN IP4 68.166.120.178 b=CT:1000 t=0 0 m=audio 23246 RTP/AVP 0 8 4 101 a=rtcp:20521 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 0: SIP/2.0 200 OK (14) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 1: Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4b K1625db0f;rport=5028 (68) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 2: From: "testcaller" ;tag=as6bb15a36 (68) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 3: To: ;tag=7a11619fdc42 4bf2885ca1c154c560a1 (68) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 4: Call-ID: 4f385fa36cff57f332bf25175d2bd33e@69.90. 155.71:5060 (59) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 5: CSeq: 102 INVITE (16) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 6: Record-Route: (59) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 7: Contact: (35) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 8: User-Agent: RTC/1.3 (19) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 9: Content-Type: application/sdp (29) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 10: Content-Length: 255 (19) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4434 parse_request: Header 11: (0) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: v=0 (3) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: o=- 0 0 IN IP4 68.166.120.178 (29) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: s=session (9) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: c=IN IP4 68.166.120.178 (23) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: b=CT:1000 (9) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: t=0 0 (5) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: m=audio 23246 RTP/AVP 0 8 4 101 (31) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: a=rtcp:20521 (12) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4466 parse_request: Line: a=fmtp:101 0-16 (15) --- (11 headers 13 lines) --- XXXX ---- this used to fail before my small code change, the strcmp of the two caller IDs would fail because one ID includes the port. Carefully look at the debug statement below. XXXX [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4254 find_call: = Found Their Call ID: '4f385fa36cff57f332bf25175d2bd33e@69.90 .155.71' Their Tag 7a11619fdc424bf2885ca1c154c560a1 Our tag: as6bb15a36 Our ID: '4f385fa36cff57f332bf25175d2bd33e@69.90.1 55.71:5060' [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:2018 __sip_ack: Acked pending invite 102 [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:2036 __sip_ack: Stopping retransmission on '4f385fa36cff57f332bf25175d2bd33e@6 9.90.155.71' of Request 102: Match Not Found [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:11389 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 68.166.120.178:23246 Got unsupported a:rtcp in SDP offer Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G723 for ID 4 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:4970 process_sdp: T38 state changed to 0 on channel SIP/cop.libretel.com-08b38 fb0 Capabilities: us - 0xd (g723|ulaw|alaw), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xd (g723|ulaw |alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 68.166.120.178:23246 [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:5047 process_sdp: We're settling with these formats: 0xd (g723|ulaw|alaw) [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:5054 process_sdp: We have an owner, now see if we need to change this call [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:2934 update_call_counter: Updating call counter for outgoing call [Oct 24 16:19:51] DEBUG[20269]: chan_sip.c:7759 build_route: build_route: Record-Route hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 69.90.155.71, port 5060 Transmitting (no NAT) to 69.90.155.71:5060: ACK sip:68.166.120.178:15076 SIP/2.0 Via: SIP/2.0/UDP 69.90.155.71:5028;branch=z9hG4bK2877dede;rport Route: From: "testcaller" ;tag=as6bb15a36 To: ;tag=7a11619fdc424bf2885ca1c154c560a1 Contact: Call-ID: 4f385fa36cff57f332bf25175d2bd33e@69.90.155.71 CSeq: 102 ACK User-Agent: Vivox Server Max-Forwards: 70 Content-Length: 0 --- [Oct 24 16:19:51] DEBUG[29449]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/cop.libretel.com-08b38fb0 > Channel SIP/cop.libretel.com-08b38fb0 was answered. [Oct 24 16:19:51] DEBUG[29449]: pbx.c:1767 pbx_extension_helper: Launching 'Answer' -- Executing [813@record-request:1] Answer("SIP/cop.libretel.com-08b38fb0", "") in new stack [Oct 24 16:19:51] DEBUG[29449]: pbx.c:1767 pbx_extension_helper: Launching 'Set' -- Executing [813@record-request:2] Set("SIP/cop.libretel.com-08b38fb0", "TIMEOUT(absolute)=300") in new stack -- Channel will hangup at 2006-10-24 20:24:51 UTC. [Oct 24 16:19:51] DEBUG[29449]: pbx.c:1767 pbx_extension_helper: Launching 'Playback' -- Executing [813@record-request:3] Playback("SIP/cop.libretel.com-08b38fb0", "demo-echotest") in new stack [Oct 24 16:19:51] DEBUG[29449]: channel.c:2680 set_format: Set channel SIP/cop.libretel.com-08b38fb0 to write format gsm [Oct 24 16:19:51] DEBUG[29449]: rtp.c:2555 ast_rtp_write: Ooh, format changed from unknown to ulaw [Oct 24 16:19:51] DEBUG[29449]: rtp.c:2572 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 -- Playing 'demo-echotest' (language 'en') [Oct 24 16:19:51] DEBUG[20229]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - cop.libretel.com [Oct 24 16:19:51] DEBUG[20229]: chan_sip.c:14831 sip_devicestate: Checking device state for peer cop.libretel.com [Oct 24 16:19:51] DEBUG[20229]: devicestate.c:287 do_state_change: Changing state for SIP/cop.libretel.com - state 4 (Inv alid) [Oct 24 16:19:51] DEBUG[29457]: app_queue.c:546 changethread: Device 'SIP/cop.libretel.com' changed to state '4' (Invalid ) but we don't care because they're not a member of any queue. [Oct 24 16:19:51] WARNING[29449]: rtp.c:1073 ast_rtp_read: RTP Read too short lib1*CLI> <--- SIP read from 69.90.155.71:5060 ---> BYE sip:testcaller@69.90.155.71:5028 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 69.90.155.71;branch=z9hG4bKb668.c1425a86.0 Via: SIP/2.0/UDP 68.166.120.178:15076 Max-Forwards: 16 From: ;tag=7a11619fdc424bf2885ca1c154c560a1 To: "testcaller" ;tag=as6bb15a36 Call-ID: 4f385fa36cff57f332bf25175d2bd33e@10.1.1.188:4127 CSeq: 1 BYE User-Agent: RTC/1.3 Content-Length: 0 Route: