*CLI> *CLI> set debug 4 Core debug was 0 and is now 4 *CLI> set verbose 43 Verbosity was 3 and is now 43 *CLI> set verbose 4 Verbosity was 43 and is now 4 *CLI> sip debug SIP Debugging enabled *CLI> ooh323 debug OOH323 Debugging Enabled *CLI> *CLI> *CLI> <-- SIP read from 10.3.2.111:5060: INVITE sip:7401@10.3.3.239;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK23464c029b146f25 From: ;tag=c20a9f0aa2a5d2a7 To: Contact: Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56792 INVITE User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 298 v=0 o=6111 8000 8000 IN IP4 10.3.2.111 s=SIP Call c=IN IP4 10.3.2.111 t=0 0 m=audio 5004 RTP/AVP 0 8 18 2 15 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:10 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 0: INVITE sip:7401@10.3.3.239;user=phone SIP/2.0 (45) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 1: Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK23464c029b146f25 (58) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 2: From: ;tag=c20a9f0aa2a5d2a7 (59) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 3: To: (36) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 4: Contact: (41) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 5: Call-ID: 3ff80074c77c069d@10.3.2.111 (36) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 6: CSeq: 56792 INVITE (18) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 7: User-Agent: Grandstream BT100 1.0.5.11 (38) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 8: Max-Forwards: 70 (16) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 10: Content-Type: application/sdp (29) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 11: Content-Length: 298 (19) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 12: (0) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: v=0 (3) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: o=6111 8000 8000 IN IP4 10.3.2.111 (34) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: s=SIP Call (10) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: c=IN IP4 10.3.2.111 (19) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: t=0 0 (5) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: m=audio 5004 RTP/AVP 0 8 18 2 15 101 (36) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:15 G728/8000 (21) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=ptime:10 (10) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=fmtp:101 0-11 (15) --- (12 headers 14 lines)--- Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3192 sip_alloc: Allocating new SIP dialog for 3ff80074c77c069d@10.3.2.111 - INVITE (With RTP) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:11186 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 3ff80074c77c069d@10.3.2.111 Sending to 10.3.2.111 : 5060 (non-NAT) Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:7204 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 10.3.2.111:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK23464c029b146f25;received=10.3.2.111 From: ;tag=c20a9f0aa2a5d2a7 To: ;tag=as53b3035b Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56792 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="2c400002" Content-Length: 0 --- Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #5 Scheduling destruction of call '3ff80074c77c069d@10.3.2.111' in 15000 ms Found user '6111' <-- SIP read from 10.3.2.111:5060: ACK sip:7401@10.3.3.239;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK23464c029b146f25 From: ;tag=c20a9f0aa2a5d2a7 To: ;tag=as53b3035b Contact: Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56792 ACK User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Jul 10 21:57:11 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 0: ACK sip:7401@10.3.3.239;user=phone SIP/2.0 (42) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 1: Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK23464c029b146f25 (58) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 2: From: ;tag=c20a9f0aa2a5d2a7 (59) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 3: To: ;tag=as53b3035b (51) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 4: Contact: (41) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 5: Call-ID: 3ff80074c77c069d@10.3.2.111 (36) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 6: CSeq: 56792 ACK (15) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 7: User-Agent: Grandstream BT100 1.0.5.11 (38) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 8: Max-Forwards: 70 (16) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 9: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 10: Content-Length: 0 (17) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:11186 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #5 Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3ff80074c77c069d@10.3.2.111' of Response 56792: Match Found <-- SIP read from 10.3.2.111:5060: INVITE sip:7401@10.3.3.239;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK54d96e5be94276c8 From: ;tag=c20a9f0aa2a5d2a7 To: Contact: Proxy-Authorization: DIGEST username="6111", realm="asterisk", algorithm=MD5, uri="sip:7401@10.3.3.239;user=phone", nonce="2c400002", response="1127d48828c6c89bddd7b4cd65fa25e5" Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56793 INVITE User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 298 v=0 o=6111 8000 8000 IN IP4 10.3.2.111 s=SIP Call c=IN IP4 10.3.2.111 t=0 0 m=audio 5004 RTP/AVP 0 8 18 2 15 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:10 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 0: INVITE sip:7401@10.3.3.239;user=phone SIP/2.0 (45) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 1: Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK54d96e5be94276c8 (58) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 2: From: ;tag=c20a9f0aa2a5d2a7 (59) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 3: To: (36) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 4: Contact: (41) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 5: Proxy-Authorization: DIGEST username="6111", realm="asterisk", algorithm=MD5, uri="sip:7401@10.3.3.239;user=phone", nonce="2c400002", response="1127d48828c6c89bddd7b4cd65fa25e5" (177) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 6: Call-ID: 3ff80074c77c069d@10.3.2.111 (36) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 7: CSeq: 56793 INVITE (18) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 8: User-Agent: Grandstream BT100 1.0.5.11 (38) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 9: Max-Forwards: 70 (16) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 11: Content-Type: application/sdp (29) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 12: Content-Length: 298 (19) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 13: (0) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: v=0 (3) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: o=6111 8000 8000 IN IP4 10.3.2.111 (34) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: s=SIP Call (10) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: c=IN IP4 10.3.2.111 (19) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: t=0 0 (5) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: m=audio 5004 RTP/AVP 0 8 18 2 15 101 (36) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 10 21:57:12 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:15 G728/8000 (21) Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=ptime:10 (10) Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=fmtp:101 0-11 (15) --- (13 headers 14 lines)--- Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:11186 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 3ff80074c77c069d@10.3.2.111 Sending to 10.3.2.111 : 5060 (non-NAT) Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:7204 check_user_full: Setting NAT on RTP to 0 Found user '6111' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 15 Found RTP audio format 101 Peer audio RTP is at port 10.3.2.111:5004 Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:3649 process_sdp: Peer audio RTP is at port 10.3.2.111:5004 Found description format PCMU Found description format PCMA Found description format G729 Found description format G726-32 Found description format G728 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x11c (ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:10546 handle_request_invite: Checking SIP call limits for device 6111 Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:2209 update_call_counter: Updating call counter for incoming call Looking for 7401 in from-internal (domain 10.3.3.239) Jul 10 21:57:13 DEBUG[7001]: chan_sip.c:6186 build_route: build_route: Contact hop: list_route: hop: Transmitting (no NAT) to 10.3.2.111:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK54d96e5be94276c8;received=10.3.2.111 From: ;tag=c20a9f0aa2a5d2a7 To: Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56793 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 10 21:57:13 DEBUG[6995]: chan_sip.c:11717 sip_devicestate: Checking device state for peer 6111 Jul 10 21:57:13 DEBUG[6995]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/6111-34fb' Jul 10 21:57:13 DEBUG[7010]: pbx.c:1677 pbx_extension_helper: Launching 'Dial' Jul 10 21:57:13 DEBUG[6995]: channel.c:777 channel_find_locked: <-- SIP read from 10.3.2.111:5060: INVITE sip:7401@10.3.3.239;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK54d96e5be94276c8 From: ;tag=c20a9f0aa2a5d2a7 To: Contact: Proxy-Authorization: DIGEST username="6111", realm="asterisk", algorithm=MD5, uri="sip:7401@10.3.3.239;user=phone", nonce="2c400002", response="1127d48828c6c89bddd7b4cd65fa25e5" Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56793 INVITE User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 298 v=0 o=6111 8000 8000 IN IP4 10.3.2.111 s=SIP Call c=IN IP4 10.3.2.111 t=0 0 m=audio 5004 RTP/AVP 0 8 18 2 15 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:10 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 Avoiding initial deadlock for 'SIP/6111-34fb' -- Executing Dial("SIP/6111-34fb", "OOH323/ken_op|20") in new stack --- ooh323_request - data ken_op format 0x4 (ulaw) Jul 10 21:57:13 DEBUG[6995]: devicestate.c:187 do_state_change: Changing state for SIP/6111 - state 2 (In use) Jul 10 21:57:13 DEBUG[6995]: chan_sip.c:11717 sip_devicestate: Checking device state for peer 6111 Jul 10 21:57:13 DEBUG[7010]: src/chan_h323.c:358 ooh323_alloc: --- ooh323_alloc Jul 10 21:57:13 DEBUG[7010]: src/chan_h323.c:441 ooh323_alloc: +++ ooh323_alloc Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 0: INVITE sip:7401@10.3.3.239;user=phone SIP/2.0 (45) --- find_peer Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 1: Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK54d96e5be94276c8 (58) Jul 10 21:57:14 DEBUG[7011]: app_queue.c:523 changethread: Device 'SIP/6111' changed to state '2' (In use) but we don't care because they're not a member of any queue. +++ find_peer Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 2: From: ;tag=c20a9f0aa2a5d2a7 (59) Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 3: To: (36) Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 4: Contact: (41) Jul 10 21:57:14 DEBUG[7010]: src/chan_h323.c:240 ooh323_new: --- ooh323_new - ken_op Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 5: Proxy-Authorization: DIGEST username="6111", realm="asterisk", algorithm=MD5, uri="sip:7401@10.3.3.239;user=phone", nonce="2c400002", response="1127d48828c6c89bddd7b4cd65fa25e5" (177) Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 6: Call-ID: 3ff80074c77c069d@10.3.2.111 (36) Jul 10 21:57:14 DEBUG[7010]: src/chan_h323.c:346 ooh323_new: +++ h323_new +++ ooh323_request Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 7: CSeq: 56793 INVITE (18) Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 8: User-Agent: Grandstream BT100 1.0.5.11 (38) Jul 10 21:57:14 DEBUG[7010]: channel.c:2829 ast_channel_inherit_variables: Not copying variable STACK-from-internal-7401-1. Jul 10 21:57:14 DEBUG[7010]: channel.c:2829 ast_channel_inherit_variables: Not copying variable SIPCALLID. Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 9: Max-Forwards: 70 (16) Jul 10 21:57:14 DEBUG[7010]: channel.c:2829 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 11: Content-Type: application/sdp (29) Jul 10 21:57:14 DEBUG[7010]: channel.c:2829 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 12: Content-Length: 298 (19) Jul 10 21:57:14 DEBUG[7010]: channel.c:2829 ast_channel_inherit_variables: Not copying variable SIPURI. --- ooh323_call- ken_op Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 13: (0) Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: v=0 (3) --- onNewCallCreated ooh323c_o_1 --- find_call Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: o=6111 8000 8000 IN IP4 10.3.2.111 (34) Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: s=SIP Call (10) +++ ooh323_call -- Called ken_op Jul 10 21:57:14 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: c=IN IP4 10.3.2.111 (19) +++ find_call setting callid number 6111 Outgoing call ken_op(ooh323c_o_1) - Codec prefs - (ulaw) -- ast_channel_make_compatible: src=0x4: dst=0x4uest: Line: t=0 0 (5) Jul 10 2Adding capabilities to call(outgoing, ooh323c_o_1) Set channel OOH323/ken_op-5800 to read format ulaw Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: m=audio 5004 RTP/AVP 0 8 18 2 15 101 (36) Jul 10 21:57:15 DEBUG[7010]: channel.c:2350 set_format: Set channel SIP/6111-34fb to write format ulaw Jul 10 21:57:15 DEBUG[7010]: channel.c:2350 set_format: Set channel SIP/6111-34fb to read format ulaw Jul 10 21:57:15 DEBUG[7010]: channel.c:2350 set_format: Set channel OOH323/ken_op-5800 to write format ulaw Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:18 G729/8000 (21) Jul 10 2Adding g711 ulaw capability to call(outgoing, ooh323c_o_1) Line: a=rtpmap:2 G726-32/8000 (23) --- configure_local_rtp Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:15 G728/8000 (21) +++ configure_local_rtp Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=ptime:10 (10) +++ onNewCallCreated ooh323c_o_1 Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=fmtp:101 0-11 (15) --- (13 headers 14 lines)--- Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:11186 handle_request: **** Received INVITE (5) - Command in SIP INVITE Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:11200 handle_request: Ignoring SIP message because of retransmit (INVITE Seqno 56793, ours 56793) Ignoring this INVITE request Jul 10 21:57:15 DEBUG[7001]: chan_sip.c:10600 handle_request_invite: Got a SIP re-invite for call 3ff80074c77c069d@10.3.2.111 Transmitting (no NAT) to 10.3.2.111:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK54d96e5be94276c8;received=10.3.2.111 From: ;tag=c20a9f0aa2a5d2a7 To: Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56793 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- --- onAlerting ooh323c_o_1 --- find_call +++ find_call Jul 10 21:57:16 DEBUG[6995]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'OOH323/ken_op-5800' -- OOH323/ken_op-5800 is ringing Transmitting (no NAT) to 10.3.2.111:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK54d96e5be94276c8;received=10.3.2.111 From: ;tag=c20a9f0aa2a5d2a7 To: ;tag=as5442c306 Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56793 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Jul 10 21:57:16 DEBUG[6995]: devicestate.c:187 do_state_change: Changing state for OOH323/ken_op - state 6 (Ringing) Jul 10 21:57:16 DEBUG[7012]: app_queue.c:523 changethread: Device 'OOH323/ken_op' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. +++ onAlerting ooh323c_o_1 *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> *CLI> --- onCallEstablished ooh323c_o_1 --- find_call +++ find_call Jul 10 21:57:20 DEBUG[6995]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'OOH323/ken_op-5800' -- OOH323/ken_op-5800 answered SIP/6111-34fb -- ast_channel_make_compatible: src=0x4: dst=0x4 Jul 10 21:57:20 DEBUG[6995]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'OOH323/ken_op-5800' +++ onCallEstablished ooh323c_o_1 Jul 10 21:57:20 DEBUG[6995]: devicestate.c:187 do_state_change: Changing state for OOH323/ken_op - state 2 (In use) Jul 10 21:57:20 DEBUG[7010]: channel.c:2350 set_format: Set channel SIP/6111-34fb to read format ulaw Jul 10 21:57:20 DEBUG[7013]: app_queue.c:523 changethread: Device 'OOH323/ken_op' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 10 21:57:20 DEBUG[7010]: channel.c:2350 set_format: Set channel OOH323/ken_op-5800 to write format ulaw Jul 10 21:57:20 DEBUG[7010]: channel.c:2350 set_format: Set channel OOH323/ken_op-5800 to read format ulaw Jul 10 21:57:20 DEBUG[7010]: channel.c:2350 set_format: Set channel SIP/6111-34fb to write format ulaw Jul 10 21:57:20 DEBUG[6995]: chan_sip.c:11717 sip_devicestate: Checking device state for peer 6111 Jul 10 21:57:20 DEBUG[6995]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/6111-34fb' Jul 10 21:57:20 DEBUG[7010]: chan_sip.c:2540 sip_answer: sip_answer(SIP/6111-34fb) We're at 10.3.3.239 port 15170 Jul 10 21:57:20 DEBUG[6995]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/6111-34fb' Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.3.2.111:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK54d96e5be94276c8;received=10.3.2.111 From: ;tag=c20a9f0aa2a5d2a7 To: ;tag=as5442c306 Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56793 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 210 v=0 o=root 7010 7010 IN IP4 10.3.3.239 s=session c=IN IP4 10.3.3.239 t=0 0 m=audio 15170 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 10 21:57:20 DEBUG[6995]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/6111-34fb' Jul 10 21:57:21 DEBUG[6995]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/6111-34fb' Jul 10 21:57:21 DEBUG[7010]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #7 -- ast_channel_bridge is being called: (1) flags=0x0 Jul 10 21:57:21 DEBUG[6995]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/6111-34fb' <-- SIP read from 10.3.2.111:5060: ACK sip:7401@10.3.3.239 SIP/2.0 Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK9198e251e7667c77 From: ;tag=c20a9f0aa2a5d2a7 To: ;tag=as5442c306 Contact: Proxy-Authorization: DIGEST username="6111", realm="asterisk", algorithm=MD5, uri="sip:7401@10.3.3.239", nonce="2c400002", response="bc22d2665c947ecf4d49cf66af15e438" Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 56793 ACK User-Agent: Grandstream BT100 1.0.5.11 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 Jul 10 21:57:21 DEBUG[6995]: devicestate.c:187 do_state_change: Changing state for SIP/6111 - state 2 (In use) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 0: ACK sip:7401@10.3.3.239 SIP/2.0 (31) -- ast_channel_bridge is being called: (2) c0-bridge=268845316: c1-bridge=268845316 Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 1: Via: SIP/2.0/UDP 10.3.2.111;branch=z9hG4bK9198e251e7667c77 (58) Jul 10 21:57:21 DEBUG[7014]: app_queue.c:523 changethread: Device 'SIP/6111' changed to state '2' (In use) but we don't care because they're not a member of any queue. Jul 10 21:57:21 DEBUG[6995]: chan_sip.c:11717 sip_devicestate: Checking device state for peer 6111 -- Attempting native bridge of SIP/6111-34fb and OOH323/ken_op-5800 Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 2: From: ;tag=c20a9f0aa2a5d2a7 (59) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 3: To: ;tag=as5442c306 (51) Jul 10 21:57:21 DEBUG[7010]: chan_sip.c:12977 sip_set_rtp_peer: Deferring reinvite on SIP '3ff80074c77c069d@10.3.2.111' - It's audio will be redirected to IP 0.0.0.0 --- ooh323_set_peer - OOH323/ken_op-5800 Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 4: Contact: (41) Jul 10 21:57:21 DEBUG[7010]: channel.c:1957 ast_read: Dropping duplicate answer! Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 5: Proxy-Authorization: DIGEST username="6111", realm="asterisk", algorithm=MD5, uri="sip:7401@10.3.3.239", nonce="2c400002", response="bc22d2665c947ecf4d49cf66af15e438" (166) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 6: Call-ID: 3ff80074c77c069d@10.3.2.111 (36) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 7: CSeq: 56793 ACK (15) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 8: User-Agent: Grandstream BT100 1.0.5.11 (38) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 9: Max-Forwards: 70 (16) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 10: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 11: Content-Length: 0 (17) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 12: (0) --- (12 headers 0 lines)--- Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:11186 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3ff80074c77c069d@10.3.2.111' of Response 56793: Match Found Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:9616 check_pendings: Sending pending reinvite on '3ff80074c77c069d@10.3.2.111' set_destination: Parsing for address/port to send to set_destination: set destination to 10.3.2.111, port 5060 X-asterisk-info: redirect=0x0: peer=0xa03026f We're at 10.3.3.239 port 15170 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 0: INVITE sip:6111@10.3.2.111 SIP/2.0 (34) Jul 10 21:57:21 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 1: Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK5cad4bb9 (55) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 2: From: ;tag=as5442c306 (53) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 3: To: ;tag=c20a9f0aa2a5d2a7 (57) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 4: Contact: (30) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 5: Call-ID: 3ff80074c77c069d@10.3.2.111 (36) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 7: User-Agent: Asterisk PBX (24) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 8: Max-Forwards: 70 (16) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (45) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 11: Content-Type: application/sdp (29) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 12: Content-Length: 210 (19) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 13: (0) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: v=0 (3) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: o=root 7010 7011 IN IP4 10.3.3.239 (34) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: s=session (9) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: c=IN IP4 10.3.3.239 (19) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: t=0 0 (5) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: m=audio 15170 RTP/AVP 0 101 (27) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=fmtp:101 0-16 (15) Jul 10 21:57:22 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 10 lines Reliably Transmitting (no NAT) to 10.3.2.111:5060: INVITE sip:6111@10.3.2.111 SIP/2.0 Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK5cad4bb9 From: ;tag=as5442c306 To: ;tag=c20a9f0aa2a5d2a7 Contact: Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 210 v=0 o=root 7010 7011 IN IP4 10.3.3.239 s=session c=IN IP4 10.3.3.239 t=0 0 m=audio 15170 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #8 == Forcing Marker bit, because SSRC has changed <-- SIP read from 10.3.2.111:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK5cad4bb9 From: ;tag=as5442c306 To: ;tag=c20a9f0aa2a5d2a7 Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 102 INVITE User-Agent: Grandstream BT100 1.0.5.11 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 195 v=0 o=6111 8000 8000 IN IP4 10.3.2.111 s=SIP Call c=IN IP4 10.3.2.111 t=0 0 m=audio 5004 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 0: SIP/2.0 200 OK (14) Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 1: Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK5cad4bb9 (55) Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 2: From: ;tag=as5442c306 (53) Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 3: To: ;tag=c20a9f0aa2a5d2a7 (57) Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 4: Call-ID: 3ff80074c77c069d@10.3.2.111 (36) Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 5: CSeq: 102 INVITE (16) --- setup_rtp_connection Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 6: User-Agent: Grandstream BT100 1.0.5.11 (38) Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 7: Contact: (41) --- find_call Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 9: Content-Type: application/sdp (29) +++ find_call Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 10: Content-Length: 195 (19) +++ setup_rtp_connection Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 11: (0) Jul 10 21:57:23 DEBUG[7010]: rtp.c:1378 ast_rtp_write: Ooh, format changed from unknown to ulaw Jul 10 21:57:23 DEBUG[7010]: rtp.c:1733 ast_rtp_bridge: Oooh, 'OOH323/ken_op-5800' changed end address to 10.1.1.155:1720 (format 0) Jul 10 21:57:23 DEBUG[7010]: rtp.c:1735 ast_rtp_bridge: Oooh, 'OOH323/ken_op-5800' changed end vaddress to 0.0.0.0:0 (format 0) Jul 10 21:57:23 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: v=0 (3) *CLI> *CLI> *CLI> *CLI> *CLI> Jul 10 21:57:24 DEBUG[7010]: rtp.c:1737 ast_rtp_bridge: Oooh, 'OOH323/ken_op-5800' was 0.0.0.0:0/(format 0) Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: o=6111 8000 8000 IN IP4 10.3.2.111 (34) Jul 10 21:57:28 DEBUG[7010]: rtp.c:1739 ast_rtp_bridge: Oooh, 'OOH323/ken_op-5800' was 0.0.0.0:0/(format 0) Jul 10 21:57:28 DEBUG[7010]: chan_sip.c:12977 sip_set_rtp_peer: Deferring reinvite on SIP '3ff80074c77c069d@10.3.2.111' - It's audio will be redirected to IP 10.1.1.155 Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: s=SIP Call (10) Jul 10 21:57:28 DEBUG[7010]: rtp.c:1378 ast_rtp_write: Ooh, format changed from unknown to ulaw Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: c=IN IP4 10.3.2.111 (19) Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: t=0 0 (5) Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: m=audio 5004 RTP/AVP 0 101 (26) Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=ptime:20 (10) Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=fmtp:101 0-11 (15) --- (11 headers 10 lines)--- Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #8 Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3ff80074c77c069d@10.3.2.111' of Request 102: Match Found Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:9631 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3ff80074c77c069d@10.3.2.111 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.3.2.111:5004 Jul 10 21:57:28 DEBUG[7001]: chan_sip.c:3649 process_sdp: Peer audio RTP is at port 10.3.2.111:5004 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:6186 build_route: build_route: Contact hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.3.2.111, port 5060 Transmitting (no NAT) to 10.3.2.111:5060: ACK sip:6111@10.3.2.111;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK61b9caff From: ;tag=as5442c306 To: ;tag=c20a9f0aa2a5d2a7 Contact: Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:9616 check_pendings: Sending pending reinvite on '3ff80074c77c069d@10.3.2.111' set_destination: Parsing for address/port to send to set_destination: set destination to 10.3.2.111, port 5060 X-asterisk-info: redirect=0xa01019b: peer=0xa03026f We're at 10.3.3.239 port 15170 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 0: INVITE sip:6111@10.3.2.111;user=phone SIP/2.0 (45) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 1: Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK0ba6a229 (55) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 2: From: ;tag=as5442c306 (53) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 3: To: ;tag=c20a9f0aa2a5d2a7 (57) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 4: Contact: (30) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 5: Call-ID: 3ff80074c77c069d@10.3.2.111 (36) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 6: CSeq: 103 INVITE (16) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 7: User-Agent: Asterisk PBX (24) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 8: Max-Forwards: 70 (16) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (45) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 11: Content-Type: application/sdp (29) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 12: Content-Length: 209 (19) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 13: (0) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: v=0 (3) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: o=root 7010 7012 IN IP4 10.1.1.155 (34) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: s=session (9) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: c=IN IP4 10.1.1.155 (19) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: t=0 0 (5) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: m=audio 1720 RTP/AVP 0 101 (26) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=fmtp:101 0-16 (15) Jul 10 21:57:29 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=silenceSupp:off - - - - (25) 13 headers, 10 lines Reliably Transmitting (no NAT) to 10.3.2.111:5060: INVITE sip:6111@10.3.2.111;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK0ba6a229 From: ;tag=as5442c306 To: ;tag=c20a9f0aa2a5d2a7 Contact: Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 209 v=0 o=root 7010 7012 IN IP4 10.1.1.155 s=session c=IN IP4 10.1.1.155 t=0 0 m=audio 1720 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #9 <-- SIP read from 10.3.2.111:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK0ba6a229 From: ;tag=as5442c306 To: ;tag=c20a9f0aa2a5d2a7 Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 103 INVITE User-Agent: Grandstream BT100 1.0.5.11 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 195 v=0 o=6111 8000 8000 IN IP4 10.3.2.111 s=SIP Call c=IN IP4 10.3.2.111 t=0 0 m=audio 5004 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 0: SIP/2.0 200 OK (14) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 1: Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK0ba6a229 (55) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 2: From: ;tag=as5442c306 (53) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 3: To: ;tag=c20a9f0aa2a5d2a7 (57) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 4: Call-ID: 3ff80074c77c069d@10.3.2.111 (36) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 5: CSeq: 103 INVITE (16) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 6: User-Agent: Grandstream BT100 1.0.5.11 (38) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 7: Contact: (41) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 9: Content-Type: application/sdp (29) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 10: Content-Length: 195 (19) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3408 parse_request: Header 11: (0) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: v=0 (3) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: o=6111 8000 8000 IN IP4 10.3.2.111 (34) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: s=SIP Call (10) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: c=IN IP4 10.3.2.111 (19) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: t=0 0 (5) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: m=audio 5004 RTP/AVP 0 101 (26) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=ptime:20 (10) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3440 parse_request: Line: a=fmtp:101 0-11 (15) --- (11 headers 10 lines)--- Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:1379 __sip_ack: Acked pending invite 103 Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #9 Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3ff80074c77c069d@10.3.2.111' of Request 103: Match Found Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:9631 handle_response_invite: SIP response 200 to RE-invite on outgoing call 3ff80074c77c069d@10.3.2.111 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.3.2.111:5004 Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:3649 process_sdp: Peer audio RTP is at port 10.3.2.111:5004 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 10 21:57:30 DEBUG[7001]: chan_sip.c:6129 build_route: build_route: Retaining previous route: set_destination: Parsing for address/port to send to set_destination: set destination to 10.3.2.111, port 5060 Transmitting (no NAT) to 10.3.2.111:5060: ACK sip:6111@10.3.2.111;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.3.3.239:5060;branch=z9hG4bK4d0fe23d From: ;tag=as5442c306 To: ;tag=c20a9f0aa2a5d2a7 Contact: Call-ID: 3ff80074c77c069d@10.3.2.111 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 10 21:57:35 DEBUG[7010]: rtp.c:415 ast_rtcp_read: Got RTCP report of 96 bytes *CLI> rtp debug RTP Debugging Enabled Got RTP packet from 10.1.1.155:1720 (type 0, seq 13016, ts 132720, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20185, ts 132480, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13017, ts 132960, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20186, ts 132640, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20187, ts 132800, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13018, ts 133200, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20188, ts 132960, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13019, ts 133440, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20189, ts 133120, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20190, ts 133280, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13020, ts 133680, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20191, ts 133440, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13021, ts 133920, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20192, ts 133600, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20193, ts 133760, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13022, ts 134160, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20194, ts 133920, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13023, ts 134400, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20195, ts 134080, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20196, ts 134240, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13024, ts 134640, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20197, ts 134400, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13025, ts 134880, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20198, ts 134560, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20199, ts 134720, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13026, ts 135120, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20200, ts 134880, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13027, ts 135360, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20201, ts 135040, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20202, ts 135200, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13028, ts 135600, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20203, ts 135360, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13029, ts 135840, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20204, ts 135520, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20205, ts 135680, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13030, ts 136080, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20206, ts 135840, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13031, ts 136320, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20207, ts 136000, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20208, ts 136160, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13032, ts 136560, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20209, ts 136320, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13033, ts 136800, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20210, ts 136480, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20211, ts 136640, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13034, ts 137040, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20212, ts 136800, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13035, ts 137280, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20213, ts 136960, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20214, ts 137120, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13036, ts 137520, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20215, ts 137280, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13037, ts 137760, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20216, ts 137440, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20217, ts 137600, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13038, ts 138000, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20218, ts 137760, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13039, ts 138240, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20219, ts 137920, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20220, ts 138080, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13040, ts 138480, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20221, ts 138240, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13041, ts 138720, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20222, ts 138400, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20223, ts 138560, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13042, ts 138960, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20224, ts 138720, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13043, ts 139200, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20225, ts 138880, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20226, ts 139040, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13044, ts 139440, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20227, ts 139200, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13045, ts 139680, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20228, ts 139360, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20229, ts 139520, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13046, ts 139920, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20230, ts 139680, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13047, ts 140160, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20231, ts 139840, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20232, ts 140000, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13048, ts 140400, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20233, ts 140160, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13049, ts 140640, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20234, ts 140320, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20235, ts 140480, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13050, ts 140880, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20236, ts 140640, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13051, ts 141120, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20237, ts 140800, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20238, ts 140960, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13052, ts 141360, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20239, ts 141120, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13053, ts 141600, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20240, ts 141280, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20241, ts 141440, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13054, ts 141840, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20242, ts 141600, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13055, ts 142080, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20243, ts 141760, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20244, ts 141920, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13056, ts 142320, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20245, ts 142080, len 160) Got RTP packet from 10.1.1.155:1720 (type 0, seq 13057, ts 142560, len 240) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20246, ts 142240, len 160) Sent RTP packet to 10.3.2.111:5004 (type 0, seq 20247, ts 142400, len 160) RTP Debugging Disabled *CLI> Jul 10 21:57:48 DEBUG[7010]: rtp.c:415 ast_rtcp_read: Got RTCP report of 96 bytes *CLI> Jul 10 21:58:03 DEBUG[7010]: rtp.c:415 ast_rtcp_read: Got RTCP report of 96 bytes set debug 0 Core debug is now OFF *CLI> set verbose 0 Verbosity is now OFF *CLI> sip no debug SIP Debugging Disabled *CLI> ooh323 no debug OOH323 Debugging Disabled *CLI>