<--- SIP read from 192.168.1.212:5060 ---> INVITE sip:201@192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK4C03A555294306C0A27E76ED71914380;rport From: "200" ;tag=1163673819 To: Contact: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 User-Agent: Kapanga Softphone Desktop 1.00/2152 Supported: timer, replaces CSeq: 1 INVITE Max-Forwards: 70 Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Length: 279 v=0 o=200 1163667398 1163673818 IN IP4 192.168.1.212 s=Kapanga [1163667398] c=IN IP4 192.168.1.212 t=0 0 m=audio 5144 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=sendrecv a=silenceSupp:off - - - a=maxptime:20 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,36 <-------------> [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 0: INVITE sip:201@192.168.1.156 SIP/2.0 (36) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK4C03A555294306C0A27E76ED71914380;rport (88) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 2: From: "200" ;tag=1163673819 (50) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 3: To: (27) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 4: Contact: (51) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (55) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 6: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 7: Supported: timer, replaces (26) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 8: CSeq: 1 INVITE (14) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 10: Session-Expires: 1800;refresher=uac (35) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 11: Content-Type: application/sdp (29) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 12: Content-Length: 279 (19) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 13: (0) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: v=0 (3) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: o=200 1163667398 1163673818 IN IP4 192.168.1.212 (48) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: s=Kapanga [1163667398] (22) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: c=IN IP4 192.168.1.212 (22) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: m=audio 5144 RTP/AVP 8 101 (26) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=sendrecv (10) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=silenceSupp:off - - - (23) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=maxptime:20 (13) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=ptime:20 (10) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=fmtp:101 0-15,36 (18) [Nov 16 11:44:20] VERBOSE[3749] logger.c: --- (13 headers 13 lines) --- [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Setting NAT on RTP to Off [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Allocating new SIP dialog for 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 - INVITE (With RTP) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Begin: parsing SIP "Supported: timer, replaces" [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Found SIP option: -timer- [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Matched SIP option: timer [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Found SIP option: -replaces- [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Matched SIP option: replaces [Nov 16 11:44:20] VERBOSE[3749] logger.c: Sending to 192.168.1.212 : 5060 (NAT) [Nov 16 11:44:20] VERBOSE[3749] logger.c: Using INVITE request as basis request - 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Setting NAT on RTP to Off [Nov 16 11:44:20] VERBOSE[3749] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.1.212:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK4C03A555294306C0A27E76ED71914380;received=192.168.1.212;rport=5060 From: "200" ;tag=1163673819 To: ;tag=as61b11d48 Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="065d94ce" Content-Length: 0 <------------> [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #916 [Nov 16 11:44:20] VERBOSE[3749] logger.c: Scheduling destruction of SIP dialog '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' in 32000 ms (Method: INVITE) [Nov 16 11:44:20] VERBOSE[3749] logger.c: Found user '200' [Nov 16 11:44:20] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.212:5060 ---> ACK sip:201@192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK4C03A555294306C0A27E76ED71914380;rport From: "200" ;tag=1163673819 To: ;tag=as61b11d48 Contact: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="065d94ce",response="a7b7b2c15cfc5b1f2aedcdd00250242a",uri="sip:201@192.168.1.156",algorithm=MD5 User-Agent: Kapanga Softphone Desktop 1.00/2152 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 0: ACK sip:201@192.168.1.156 SIP/2.0 (33) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK4C03A555294306C0A27E76ED71914380;rport (88) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 2: From: "200" ;tag=1163673819 (50) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 3: To: ;tag=as61b11d48 (42) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 4: Contact: (51) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (55) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 6: Proxy-Authorization: Digest username="200",realm="asterisk",nonce="065d94ce",response="a7b7b2c15cfc5b1f2aedcdd00250242a",uri="sip:201@192.168.1.156",algorithm=MD5 (162) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 7: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 8: CSeq: 1 ACK (11) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 10: Content-Length: 0 (17) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 11: (0) [Nov 16 11:44:20] VERBOSE[3749] logger.c: --- (11 headers 0 lines) --- [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #916 [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Stopping retransmission on '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' of Response 1: Match Not Found [Nov 16 11:44:20] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.212:5060 ---> INVITE sip:201@192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bKEF6F7EB63D68338AC3D29ACBB89E9E27;rport From: "200" ;tag=1163673819 To: Contact: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="065d94ce",response="a7b7b2c15cfc5b1f2aedcdd00250242a",uri="sip:201@192.168.1.156",algorithm=MD5 User-Agent: Kapanga Softphone Desktop 1.00/2152 Supported: timer, replaces CSeq: 2 INVITE Max-Forwards: 70 Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Length: 279 v=0 o=200 1163667398 1163673819 IN IP4 192.168.1.212 s=Kapanga [1163667398] c=IN IP4 192.168.1.212 t=0 0 m=audio 5144 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=sendrecv a=silenceSupp:off - - - a=maxptime:20 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,36 <-------------> [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 0: INVITE sip:201@192.168.1.156 SIP/2.0 (36) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bKEF6F7EB63D68338AC3D29ACBB89E9E27;rport (88) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 2: From: "200" ;tag=1163673819 (50) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 3: To: (27) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 4: Contact: (51) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (55) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 6: Proxy-Authorization: Digest username="200",realm="asterisk",nonce="065d94ce",response="a7b7b2c15cfc5b1f2aedcdd00250242a",uri="sip:201@192.168.1.156",algorithm=MD5 (162) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 7: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 8: Supported: timer, replaces (26) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 9: CSeq: 2 INVITE (14) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 10: Max-Forwards: 70 (16) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 11: Session-Expires: 1800;refresher=uac (35) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 13: Content-Length: 279 (19) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 14: (0) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: v=0 (3) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: o=200 1163667398 1163673819 IN IP4 192.168.1.212 (48) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: s=Kapanga [1163667398] (22) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: c=IN IP4 192.168.1.212 (22) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: m=audio 5144 RTP/AVP 8 101 (26) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=rtpmap:8 pcma/8000 (20) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=sendrecv (10) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=silenceSupp:off - - - (23) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=maxptime:20 (13) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=ptime:20 (10) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Line: a=fmtp:101 0-15,36 (18) [Nov 16 11:44:20] VERBOSE[3749] logger.c: --- (14 headers 13 lines) --- [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 16 11:44:20] VERBOSE[3749] logger.c: Sending to 192.168.1.212 : 5060 (NAT) [Nov 16 11:44:20] VERBOSE[3749] logger.c: Using INVITE request as basis request - 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Setting NAT on RTP to Off [Nov 16 11:44:20] VERBOSE[3749] logger.c: Found user '200' [Nov 16 11:44:20] VERBOSE[3749] logger.c: Found RTP audio format 8 [Nov 16 11:44:20] VERBOSE[3749] logger.c: Found RTP audio format 101 [Nov 16 11:44:20] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.212:5144 [Nov 16 11:44:20] VERBOSE[3749] logger.c: Found description format pcma for ID 8 [Nov 16 11:44:20] VERBOSE[3749] logger.c: Found description format telephone-event for ID 101 [Nov 16 11:44:20] VERBOSE[3749] logger.c: Got unsupported a:fmtp in SDP offer [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: T38 state changed to 0 on channel [Nov 16 11:44:20] VERBOSE[3749] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Nov 16 11:44:20] VERBOSE[3749] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 16 11:44:20] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.212:5144 [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Checking SIP call limits for device 200 [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Updating call counter for incoming call [Nov 16 11:44:20] VERBOSE[3749] logger.c: Looking for 201 in from-sip (domain 192.168.1.156) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: *** Our capabilities are 0x8 (alaw) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: This channel will not be able to handle video. [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: build_route: Contact hop: [Nov 16 11:44:20] VERBOSE[3749] logger.c: list_route: hop: [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: SIP/200-088020c8: New call is still down.... Trying... [Nov 16 11:44:20] VERBOSE[3749] logger.c: <--- Transmitting (no NAT) to 192.168.1.212:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bKEF6F7EB63D68338AC3D29ACBB89E9E27;received=192.168.1.212;rport=5060 From: "200" ;tag=1163673819 To: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Nov 16 11:44:20] DEBUG[3749] devicestate.c: Notification of state change to be queued on device/channel SIP/200-088020c8 [Nov 16 11:44:20] DEBUG[8476] pbx.c: Launching 'Dial' [Nov 16 11:44:20] VERBOSE[8476] logger.c: -- Executing [201@from-sip:1] Dial("SIP/200-088020c8", "SIP/201|300") in new stack [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Asked to create a SIP channel with formats: 0x8 (alaw) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Setting NAT on RTP to Off [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: *** Our native formats are 0x8 (alaw) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: *** Our capabilities are 0x8 (alaw) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: *** Our preferred formats from the incoming channel are 0x8 (alaw) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: This channel will not be able to handle video. [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Early remote bridge setting SIP '7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156' - Sending media to 192.168.1.212 [Nov 16 11:44:20] DEBUG[8476] rtp.c: Seeded SDP of 'SIP/201-08808278' with that of 'SIP/200-088020c8' [Nov 16 11:44:20] DEBUG[8476] channel.c: Not copying variable STACK-from-sip-201-1. [Nov 16 11:44:20] DEBUG[8476] channel.c: Not copying variable SIPCALLID. [Nov 16 11:44:20] DEBUG[8476] channel.c: Not copying variable SIPUSERAGENT. [Nov 16 11:44:20] DEBUG[8476] channel.c: Not copying variable SIPDOMAIN. [Nov 16 11:44:20] DEBUG[8476] channel.c: Not copying variable SIPURI. [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Outgoing Call for 201 [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Updating call counter for outgoing call [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Our T38 capability (3840), joint T38 capability (3840) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: False [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 16 11:44:20] VERBOSE[8476] logger.c: Audio is at 192.168.1.156 port 15888 [Nov 16 11:44:20] VERBOSE[8476] logger.c: Adding codec 0x8 (alaw) to SDP [Nov 16 11:44:20] VERBOSE[8476] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: -- Done with adding codecs to SDP [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 0: INVITE sip:201@192.168.1.208:5060 SIP/2.0 (41) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK30108fa9;rport (64) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 2: From: "200" ;tag=as2e312935 (50) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 3: To: (32) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 4: Contact: (32) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 5: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 (55) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 9: Date: Thu, 16 Nov 2006 10:44:20 GMT (35) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 11: Supported: replaces (19) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 13: Content-Length: 239 (19) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Header 14: (0) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: v=0 (3) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: o=root 3728 3728 IN IP4 192.168.1.212 (37) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: s=session (9) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: c=IN IP4 192.168.1.212 (22) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: m=audio 5144 RTP/AVP 8 101 (26) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: a=ptime:20 (10) [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: Line: a=sendrecv (10) [Nov 16 11:44:20] VERBOSE[8476] logger.c: Reliably Transmitting (no NAT) to 192.168.1.208:5060: INVITE sip:201@192.168.1.208:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK30108fa9;rport From: "200" ;tag=as2e312935 To: Contact: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 16 Nov 2006 10:44:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 239 v=0 o=root 3728 3728 IN IP4 192.168.1.212 s=session c=IN IP4 192.168.1.212 t=0 0 m=audio 5144 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Nov 16 11:44:20] DEBUG[8476] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #918 [Nov 16 11:44:20] VERBOSE[8476] logger.c: -- Called 201 [Nov 16 11:44:20] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> SIP/2.0 100 Trying Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 102 INVITE From: "200" ;tag=as2e312935 To: ;tag=86878122aefef67 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK30108fa9;rport Content-Length: 0 User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 <-------------> [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 1: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 (55) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 2: CSeq: 102 INVITE (16) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 3: From: "200" ;tag=as2e312935 (50) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 4: To: ;tag=86878122aefef67 (52) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK30108fa9;rport (64) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 6: Content-Length: 0 (17) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 7: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 8: (0) [Nov 16 11:44:20] VERBOSE[3749] logger.c: --- (8 headers 0 lines) --- [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: *** SIP TIMER: Cancelling retransmission #918 - INVITE (got response) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156' Request 102: Found [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: SIP response 100 to standard invite [Nov 16 11:44:20] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> SIP/2.0 180 Ringing Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 102 INVITE From: "200" ;tag=as2e312935 To: ;tag=86878122aefef67 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK30108fa9;rport Content-Length: 0 Contact: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 <-------------> [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 1: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 (55) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 2: CSeq: 102 INVITE (16) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 3: From: "200" ;tag=as2e312935 (50) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 4: To: ;tag=86878122aefef67 (52) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK30108fa9;rport (64) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 6: Content-Length: 0 (17) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 7: Contact: (37) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 8: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: Header 9: (0) [Nov 16 11:44:20] VERBOSE[3749] logger.c: --- (9 headers 0 lines) --- [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156' Request 102: Found [Nov 16 11:44:20] DEBUG[3749] chan_sip.c: SIP response 180 to standard invite [Nov 16 11:44:20] DEBUG[3749] devicestate.c: Notification of state change to be queued on device/channel SIP/201-08808278 [Nov 16 11:44:20] VERBOSE[8476] logger.c: -- SIP/201-08808278 is ringing [Nov 16 11:44:20] DEBUG[8476] rtp.c: Setting early bridge SDP of 'SIP/200-088020c8' with that of 'SIP/201-08808278' [Nov 16 11:44:20] VERBOSE[8476] logger.c: <--- Transmitting (no NAT) to 192.168.1.212:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bKEF6F7EB63D68338AC3D29ACBB89E9E27;received=192.168.1.212;rport=5060 From: "200" ;tag=1163673819 To: ;tag=as25534657 Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Nov 16 11:44:21] DEBUG[3731] devicestate.c: No provider found, checking channel drivers for SIP - 200 [Nov 16 11:44:21] DEBUG[3731] chan_sip.c: Checking device state for peer 200 [Nov 16 11:44:21] DEBUG[3731] devicestate.c: Changing state for SIP/200 - state 1 (Not in use) [Nov 16 11:44:21] DEBUG[3731] devicestate.c: No provider found, checking channel drivers for SIP - 201 [Nov 16 11:44:21] DEBUG[3731] chan_sip.c: Checking device state for peer 201 [Nov 16 11:44:21] DEBUG[3731] devicestate.c: Changing state for SIP/201 - state 1 (Not in use) [Nov 16 11:44:21] DEBUG[8477] app_queue.c: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 16 11:44:21] DEBUG[8478] app_queue.c: Device 'SIP/201' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 16 11:44:27] DEBUG[3749] chan_sip.c: Auto destroying SIP dialog '7479F055307F434C30B8470A943785A3@192.168.1.156' [Nov 16 11:44:27] DEBUG[3749] chan_sip.c: Destroying SIP dialog 7479F055307F434C30B8470A943785A3@192.168.1.156 [Nov 16 11:44:27] VERBOSE[3749] logger.c: Really destroying SIP dialog '7479F055307F434C30B8470A943785A3@192.168.1.156' Method: OPTIONS [Nov 16 11:44:29] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> SIP/2.0 200 OK Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 102 INVITE From: "200" ;tag=as2e312935 To: ;tag=86878122aefef67 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK30108fa9;rport Content-Length: 244 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Supported: replaces Contact: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 v=0 o=MxSIP 0 817888667 IN IP4 192.168.1.208 s=SIP Call c=IN IP4 192.168.1.208 t=0 0 m=audio 10014 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - <-------------> [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 1: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 (55) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 2: CSeq: 102 INVITE (16) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 3: From: "200" ;tag=as2e312935 (50) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 4: To: ;tag=86878122aefef67 (52) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK30108fa9;rport (64) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 6: Content-Length: 244 (19) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 7: Allow: NOTIFY (13) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 8: Allow: REFER (12) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 9: Allow: OPTIONS (14) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 10: Allow: INVITE (13) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 11: Allow: ACK (10) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 12: Allow: CANCEL (13) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 13: Allow: BYE (10) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 14: Content-Type: application/sdp (29) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 15: Supported: replaces (19) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 16: Contact: (37) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 17: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 18: (0) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: v=0 (3) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: o=MxSIP 0 817888667 IN IP4 192.168.1.208 (40) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: s=SIP Call (10) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: c=IN IP4 192.168.1.208 (22) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: m=audio 10014 RTP/AVP 8 101 (27) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: a=ptime:20 (10) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: a=sendrecv (10) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Nov 16 11:44:29] VERBOSE[3749] logger.c: --- (18 headers 12 lines) --- [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Acked pending invite 102 [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Stopping retransmission on '7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156' of Request 102: Match Not Found [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: SIP response 200 to standard invite [Nov 16 11:44:29] VERBOSE[3749] logger.c: Found RTP audio format 8 [Nov 16 11:44:29] VERBOSE[3749] logger.c: Found RTP audio format 101 [Nov 16 11:44:29] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.208:10014 [Nov 16 11:44:29] VERBOSE[3749] logger.c: Found description format PCMA for ID 8 [Nov 16 11:44:29] VERBOSE[3749] logger.c: Found description format telephone-event for ID 101 [Nov 16 11:44:29] VERBOSE[3749] logger.c: Got unsupported a:fmtp in SDP offer [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: T38 state changed to 0 on channel SIP/201-08808278 [Nov 16 11:44:29] VERBOSE[3749] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Nov 16 11:44:29] VERBOSE[3749] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 16 11:44:29] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.208:10014 [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: We have an owner, now see if we need to change this call [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Updating call counter for outgoing call [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: build_route: Contact hop: [Nov 16 11:44:29] VERBOSE[3749] logger.c: list_route: hop: [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Strict routing enforced for session 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 [Nov 16 11:44:29] VERBOSE[3749] logger.c: set_destination: Parsing for address/port to send to [Nov 16 11:44:29] VERBOSE[3749] logger.c: set_destination: set destination to 192.168.1.208, port 5060 [Nov 16 11:44:29] VERBOSE[3749] logger.c: Transmitting (no NAT) to 192.168.1.208:5060: ACK sip:201@192.168.1.208:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK083b9e3b;rport From: "200" ;tag=as2e312935 To: ;tag=86878122aefef67 Contact: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 16 11:44:29] DEBUG[8476] devicestate.c: Notification of state change to be queued on device/channel SIP/201-08808278 [Nov 16 11:44:29] VERBOSE[8476] logger.c: -- SIP/201-08808278 answered SIP/200-088020c8 [Nov 16 11:44:29] DEBUG[8476] chan_sip.c: Early remote bridge setting SIP '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' - Sending media to 192.168.1.208 [Nov 16 11:44:29] DEBUG[8476] rtp.c: Setting early bridge SDP of 'SIP/200-088020c8' with that of 'SIP/201-08808278' [Nov 16 11:44:29] DEBUG[8476] devicestate.c: Notification of state change to be queued on device/channel SIP/200-088020c8 [Nov 16 11:44:29] DEBUG[8476] chan_sip.c: SIP answering channel: SIP/200-088020c8 [Nov 16 11:44:29] DEBUG[8476] chan_sip.c: Setting framing from config on incoming call [Nov 16 11:44:29] DEBUG[8476] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Nov 16 11:44:29] DEBUG[8476] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 16 11:44:29] VERBOSE[8476] logger.c: Audio is at 192.168.1.156 port 15700 [Nov 16 11:44:29] VERBOSE[8476] logger.c: Adding codec 0x8 (alaw) to SDP [Nov 16 11:44:29] VERBOSE[8476] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 16 11:44:29] DEBUG[8476] chan_sip.c: -- Done with adding codecs to SDP [Nov 16 11:44:29] DEBUG[8476] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Nov 16 11:44:29] VERBOSE[8476] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.1.212:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bKEF6F7EB63D68338AC3D29ACBB89E9E27;received=192.168.1.212;rport=5060 From: "200" ;tag=1163673819 To: ;tag=as25534657 Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 3728 3728 IN IP4 192.168.1.208 s=session c=IN IP4 192.168.1.208 t=0 0 m=audio 10014 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Nov 16 11:44:29] DEBUG[8476] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #920 [Nov 16 11:44:29] VERBOSE[8476] logger.c: -- Native bridging SIP/200-088020c8 and SIP/201-08808278 [Nov 16 11:44:29] DEBUG[3731] devicestate.c: No provider found, checking channel drivers for SIP - 201 [Nov 16 11:44:29] DEBUG[3731] chan_sip.c: Checking device state for peer 201 [Nov 16 11:44:29] DEBUG[3731] devicestate.c: Changing state for SIP/201 - state 1 (Not in use) [Nov 16 11:44:29] DEBUG[3731] devicestate.c: No provider found, checking channel drivers for SIP - 200 [Nov 16 11:44:29] DEBUG[3731] chan_sip.c: Checking device state for peer 200 [Nov 16 11:44:29] DEBUG[3731] devicestate.c: Changing state for SIP/200 - state 1 (Not in use) [Nov 16 11:44:29] DEBUG[8481] app_queue.c: Device 'SIP/201' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 16 11:44:29] DEBUG[8482] app_queue.c: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 16 11:44:29] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.212:5060 ---> ACK sip:201@192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK40AB86393DDA7A5EA87C76D29BD5E6A5;rport From: "200" ;tag=1163673819 To: ;tag=as25534657 Contact: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="065d94ce",response="a7b7b2c15cfc5b1f2aedcdd00250242a",uri="sip:201@192.168.1.156",algorithm=MD5 User-Agent: Kapanga Softphone Desktop 1.00/2152 CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 0: ACK sip:201@192.168.1.156 SIP/2.0 (33) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK40AB86393DDA7A5EA87C76D29BD5E6A5;rport (88) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 2: From: "200" ;tag=1163673819 (50) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 3: To: ;tag=as25534657 (42) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 4: Contact: (51) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (55) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 6: Proxy-Authorization: Digest username="200",realm="asterisk",nonce="065d94ce",response="a7b7b2c15cfc5b1f2aedcdd00250242a",uri="sip:201@192.168.1.156",algorithm=MD5 (162) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 7: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 8: CSeq: 2 ACK (11) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 10: Content-Length: 0 (17) [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Header 11: (0) [Nov 16 11:44:29] VERBOSE[3749] logger.c: --- (11 headers 0 lines) --- [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #920 [Nov 16 11:44:29] DEBUG[3749] chan_sip.c: Stopping retransmission on '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' of Response 2: Match Not Found [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 0: OPTIONS sip:201@192.168.1.208:5060 SIP/2.0 (42) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK6f7e07c3;rport (64) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 2: From: "asterisk" ;tag=as5d07c3de (60) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 3: To: (32) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 4: Contact: (37) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 427f5faf0b4a89803ec1a79f4d13be40@192.168.1.156 (55) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 9: Date: Thu, 16 Nov 2006 10:44:32 GMT (35) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 11: Supported: replaces (19) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 12: Content-Length: 0 (17) [Nov 16 11:44:32] VERBOSE[3749] logger.c: Reliably Transmitting (no NAT) to 192.168.1.208:5060: OPTIONS sip:201@192.168.1.208:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK6f7e07c3;rport From: "asterisk" ;tag=as5d07c3de To: Contact: Call-ID: 427f5faf0b4a89803ec1a79f4d13be40@192.168.1.156 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 16 Nov 2006 10:44:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #921 [Nov 16 11:44:32] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> SIP/2.0 200 OK Call-ID: 427f5faf0b4a89803ec1a79f4d13be40@192.168.1.156 CSeq: 102 OPTIONS From: "asterisk" ;tag=as5d07c3de To: ;tag=0f58a67e4fd80e0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK6f7e07c3;rport Content-Length: 0 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Contact: Supported: replaces User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 <-------------> [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 1: Call-ID: 427f5faf0b4a89803ec1a79f4d13be40@192.168.1.156 (55) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 2: CSeq: 102 OPTIONS (17) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 3: From: "asterisk" ;tag=as5d07c3de (60) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 4: To: ;tag=0f58a67e4fd80e0 (52) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK6f7e07c3;rport (64) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 6: Content-Length: 0 (17) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 7: Allow: NOTIFY (13) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 8: Allow: REFER (12) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 9: Allow: OPTIONS (14) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 10: Allow: INVITE (13) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 11: Allow: ACK (10) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 12: Allow: CANCEL (13) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 13: Allow: BYE (10) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 14: Contact: (37) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 15: Supported: replaces (19) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 16: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Header 17: (0) [Nov 16 11:44:32] VERBOSE[3749] logger.c: --- (17 headers 0 lines) --- [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #921 [Nov 16 11:44:32] DEBUG[3749] chan_sip.c: Stopping retransmission on '427f5faf0b4a89803ec1a79f4d13be40@192.168.1.156' of Request 102: Match Not Found [Nov 16 11:44:32] VERBOSE[3749] logger.c: Really destroying SIP dialog '427f5faf0b4a89803ec1a79f4d13be40@192.168.1.156' Method: OPTIONS [Nov 16 11:44:34] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> INVITE sip:200@192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKc5fbcf755 Max-Forwards: 70 Content-Length: 115 To: "200" ;tag=as2e312935 From: ;tag=86878122aefef67 Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 31662862 INVITE Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Contact: Supported: replaces User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 v=0 o=MxSIP 0 817888667 IN IP4 192.168.1.208 s=SIP Call c=IN IP4 192.168.1.208 t=0 0 m=image 10014 UDPTL t38 <-------------> [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 0: INVITE sip:200@192.168.1.156 SIP/2.0 (36) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKc5fbcf755 (59) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 3: Content-Length: 115 (19) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 4: To: "200" ;tag=as2e312935 (48) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 5: From: ;tag=86878122aefef67 (54) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 6: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 (55) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 7: CSeq: 31662862 INVITE (21) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 8: Supported: timer (16) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 9: Allow: NOTIFY (13) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 10: Allow: REFER (12) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 11: Allow: OPTIONS (14) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 12: Allow: INVITE (13) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 13: Allow: ACK (10) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 14: Allow: CANCEL (13) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 15: Allow: BYE (10) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 16: Content-Type: application/sdp (29) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 17: Contact: (37) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 18: Supported: replaces (19) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 19: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 20: (0) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Line: v=0 (3) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Line: o=MxSIP 0 817888667 IN IP4 192.168.1.208 (40) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Line: s=SIP Call (10) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Line: c=IN IP4 192.168.1.208 (22) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Line: m=image 10014 UDPTL t38 (23) [Nov 16 11:44:34] VERBOSE[3749] logger.c: --- (20 headers 6 lines) --- [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Begin: parsing SIP "Supported: timer" [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Found SIP option: -timer- [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Matched SIP option: timer [Nov 16 11:44:34] VERBOSE[3749] logger.c: Sending to 192.168.1.208 : 5060 (no NAT) [Nov 16 11:44:34] WARNING[3749] chan_sip.c: Unsupported SDP media type in offer: image 10014 UDPTL t38 [Nov 16 11:44:34] VERBOSE[3749] logger.c: <--- Transmitting (no NAT) to 192.168.1.208:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKc5fbcf755;received=192.168.1.208 From: ;tag=86878122aefef67 To: "200" ;tag=as2e312935 Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 31662862 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: SIP message could not be handled, bad request: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 [Nov 16 11:44:34] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> ACK sip:200@192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKc5fbcf755 Max-Forwards: 70 Content-Length: 0 To: "200" ;tag=as2e312935 From: ;tag=86878122aefef67 Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 31662862 ACK User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 <-------------> [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 0: ACK sip:200@192.168.1.156 SIP/2.0 (33) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKc5fbcf755 (59) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 3: Content-Length: 0 (17) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 4: To: "200" ;tag=as2e312935 (48) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 5: From: ;tag=86878122aefef67 (54) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 6: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 (55) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 7: CSeq: 31662862 ACK (18) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 8: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Header 9: (0) [Nov 16 11:44:34] VERBOSE[3749] logger.c: --- (9 headers 0 lines) --- [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 16 11:44:34] DEBUG[3749] chan_sip.c: Stopping retransmission on '7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156' of Response 31662862: Match Found [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 0: OPTIONS sip:200@192.168.1.212:5060;transport=udp SIP/2.0 (56) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK05182978;rport (64) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 2: From: "asterisk" ;tag=as6f8171a6 (60) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 3: To: (46) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 4: Contact: (37) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 569dc58e6e46bcab0fce23b71527ff6e@192.168.1.156 (55) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 9: Date: Thu, 16 Nov 2006 10:44:37 GMT (35) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 11: Supported: replaces (19) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 12: Content-Length: 0 (17) [Nov 16 11:44:37] VERBOSE[3749] logger.c: Reliably Transmitting (no NAT) to 192.168.1.212:5060: OPTIONS sip:200@192.168.1.212:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK05182978;rport From: "asterisk" ;tag=as6f8171a6 To: Contact: Call-ID: 569dc58e6e46bcab0fce23b71527ff6e@192.168.1.156 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 16 Nov 2006 10:44:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #924 [Nov 16 11:44:37] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.212:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK05182978;rport=5060;received=192.168.1.156 User-Agent: Kapanga Softphone Desktop 1.00/2152 From: ;tag=1163673836 To: "asterisk" ;tag=as6f8171a6 Call-ID: 569dc58e6e46bcab0fce23b71527ff6e@192.168.1.156 CSeq: 102 OPTIONS Contact: Supported: timer, replaces Content-Length: 0 <-------------> [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK05182978;rport=5060;received=192.168.1.156 (92) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 2: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 3: From: ;tag=1163673836 (63) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 4: To: "asterisk" ;tag=as6f8171a6 (58) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 569dc58e6e46bcab0fce23b71527ff6e@192.168.1.156 (55) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 7: Contact: (51) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 8: Supported: timer, replaces (26) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 9: Content-Length: 0 (17) [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Header 10: (0) [Nov 16 11:44:37] VERBOSE[3749] logger.c: --- (10 headers 0 lines) --- [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #924 [Nov 16 11:44:37] DEBUG[3749] chan_sip.c: Stopping retransmission on '569dc58e6e46bcab0fce23b71527ff6e@192.168.1.156' of Request 102: Match Not Found [Nov 16 11:44:37] VERBOSE[3749] logger.c: Really destroying SIP dialog '569dc58e6e46bcab0fce23b71527ff6e@192.168.1.156' Method: OPTIONS [Nov 16 11:44:41] DEBUG[3749] chan_sip.c: Auto destroying SIP dialog 'e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208' [Nov 16 11:44:41] DEBUG[3749] chan_sip.c: Destroying SIP dialog e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 [Nov 16 11:44:41] VERBOSE[3749] logger.c: Really destroying SIP dialog 'e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208' Method: REGISTER [Nov 16 11:44:57] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.212:5060 ---> OPTIONS sip:192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK25E8B4C331F528FF5103A86DEEC61B70;rport From: "200" ;tag=1163667389 To: "200" ;tag=as79510051 Contact: Call-ID: 7479F055307F434C30B8470A943785A3@192.168.1.156 User-Agent: Kapanga Softphone Desktop 1.00/2152 Supported: timer, replaces CSeq: 106 OPTIONS Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 0: OPTIONS sip:192.168.1.156 SIP/2.0 (33) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK25E8B4C331F528FF5103A86DEEC61B70;rport (88) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 2: From: "200" ;tag=1163667389 (50) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 3: To: "200" ;tag=as79510051 (48) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 4: Contact: (51) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 7479F055307F434C30B8470A943785A3@192.168.1.156 (55) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 6: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 7: Supported: timer, replaces (26) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 8: CSeq: 106 OPTIONS (17) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 9: Max-Forwards: 70 (16) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 10: Content-Length: 0 (17) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Header 11: (0) [Nov 16 11:44:57] VERBOSE[3749] logger.c: --- (11 headers 0 lines) --- [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: Allocating new SIP dialog for 7479F055307F434C30B8470A943785A3@192.168.1.156 - OPTIONS (No RTP) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Nov 16 11:44:57] VERBOSE[3749] logger.c: Looking for s in default (domain 192.168.1.156) [Nov 16 11:44:57] VERBOSE[3749] logger.c: <--- Transmitting (no NAT) to 192.168.1.212:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.212:5060;branch=z9hG4bK25E8B4C331F528FF5103A86DEEC61B70;received=192.168.1.212;rport=5060 From: "200" ;tag=1163667389 To: "200" ;tag=as79510051 Call-ID: 7479F055307F434C30B8470A943785A3@192.168.1.156 CSeq: 106 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 <------------> [Nov 16 11:44:57] VERBOSE[3749] logger.c: Scheduling destruction of SIP dialog '7479F055307F434C30B8470A943785A3@192.168.1.156' in 32000 ms (Method: OPTIONS) [Nov 16 11:44:57] DEBUG[3749] chan_sip.c: SIP message could not be handled, bad request: 7479F055307F434C30B8470A943785A3@192.168.1.156 [Nov 16 11:45:08] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> INVITE sip:200@192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bK0a567c4b0 Max-Forwards: 70 Content-Length: 420 To: "200" ;tag=as2e312935 From: ;tag=86878122aefef67 Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 31662863 INVITE Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Content-Type: application/sdp Contact: Supported: replaces User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 v=0 o=MxSIP 0 817888667 IN IP4 192.168.1.208 s=SIP Call c=IN IP4 192.168.1.208 t=0 0 m=audio 10014 RTP/AVP 8 0 97 98 2 99 4 18 100 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 G726-16/8000 a=rtpmap:98 G726-24/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:99 G726-40/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=silenceSupp:off - - - - <-------------> [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 0: INVITE sip:200@192.168.1.156 SIP/2.0 (36) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bK0a567c4b0 (59) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 3: Content-Length: 420 (19) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 4: To: "200" ;tag=as2e312935 (48) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 5: From: ;tag=86878122aefef67 (54) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 6: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 (55) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 7: CSeq: 31662863 INVITE (21) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 8: Supported: timer (16) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 9: Allow: NOTIFY (13) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 10: Allow: REFER (12) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 11: Allow: OPTIONS (14) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 12: Allow: INVITE (13) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 13: Allow: ACK (10) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 14: Allow: CANCEL (13) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 15: Allow: BYE (10) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 16: Content-Type: application/sdp (29) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 17: Contact: (37) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 18: Supported: replaces (19) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 19: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Header 20: (0) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: v=0 (3) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: o=MxSIP 0 817888667 IN IP4 192.168.1.208 (40) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: s=SIP Call (10) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: c=IN IP4 192.168.1.208 (22) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: m=audio 10014 RTP/AVP 8 0 97 98 2 99 4 18 100 (45) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=rtpmap:97 G726-16/8000 (24) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=rtpmap:98 G726-24/8000 (24) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=rtpmap:2 G726-32/8000 (23) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=rtpmap:99 G726-40/8000 (24) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=rtpmap:100 telephone-event/8000 (33) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=ptime:20 (10) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Nov 16 11:45:08] VERBOSE[3749] logger.c: --- (20 headers 18 lines) --- [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Nov 16 11:45:08] VERBOSE[3749] logger.c: Sending to 192.168.1.208 : 5060 (no NAT) [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found RTP audio format 8 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found RTP audio format 0 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found RTP audio format 97 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found RTP audio format 98 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found RTP audio format 2 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found RTP audio format 99 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found RTP audio format 4 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found RTP audio format 18 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found RTP audio format 100 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.208:10014 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found description format PCMA for ID 8 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found description format PCMU for ID 0 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found description format G726-16 for ID 97 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found description format G726-24 for ID 98 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found description format G726-32 for ID 2 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found description format G726-40 for ID 99 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found description format G723 for ID 4 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found description format G729 for ID 18 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Found description format telephone-event for ID 100 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Got unsupported a:fmtp in SDP offer [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: T38 state changed to 0 on channel SIP/201-08808278 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x200d0d (g723|ulaw|alaw|g726|g729|ilbc|h264)/video=0x0 (nothing), combined - 0x8 (alaw) [Nov 16 11:45:08] VERBOSE[3749] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 16 11:45:08] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.208:10014 [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: We have an owner, now see if we need to change this call [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Got a SIP re-invite for call 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: SIP/201-08808278: This call is UP.... [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Setting framing from config on incoming call [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: ** Our prefcodec: 0x8 (alaw) [Nov 16 11:45:08] VERBOSE[3749] logger.c: Audio is at 192.168.1.156 port 15888 [Nov 16 11:45:08] VERBOSE[3749] logger.c: Adding codec 0x8 (alaw) to SDP [Nov 16 11:45:08] VERBOSE[3749] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: -- Done with adding codecs to SDP [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Nov 16 11:45:08] VERBOSE[3749] logger.c: <--- Reliably Transmitting (no NAT) to 192.168.1.208:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bK0a567c4b0;received=192.168.1.208 From: ;tag=86878122aefef67 To: "200" ;tag=as2e312935 Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 31662863 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 239 v=0 o=root 3728 3729 IN IP4 192.168.1.212 s=session c=IN IP4 192.168.1.212 t=0 0 m=audio 5144 RTP/AVP 8 100 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Nov 16 11:45:08] DEBUG[3749] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #928 [Nov 16 11:45:09] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> ACK sip:200@192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bK770e995d6 Max-Forwards: 70 Content-Length: 0 To: "200" ;tag=as2e312935 From: ;tag=86878122aefef67 Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 31662863 ACK Contact: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 <-------------> [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 0: ACK sip:200@192.168.1.156 SIP/2.0 (33) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bK770e995d6 (59) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 3: Content-Length: 0 (17) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 4: To: "200" ;tag=as2e312935 (48) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 5: From: ;tag=86878122aefef67 (54) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 6: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 (55) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 7: CSeq: 31662863 ACK (18) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 8: Contact: (37) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 9: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 10: (0) [Nov 16 11:45:09] VERBOSE[3749] logger.c: --- (10 headers 0 lines) --- [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #928 [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Stopping retransmission on '7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156' of Response 31662863: Match Not Found [Nov 16 11:45:09] DEBUG[8476] rtp.c: Oooh, 'SIP/201-08808278' changed end address to 192.168.1.208:10014 (format 2100493) [Nov 16 11:45:09] DEBUG[8476] rtp.c: Oooh, 'SIP/201-08808278' changed end vaddress to 0.0.0.0:0 (format 2100493) [Nov 16 11:45:09] DEBUG[8476] rtp.c: Oooh, 'SIP/201-08808278' was 192.168.1.208:10014/(format 8) [Nov 16 11:45:09] DEBUG[8476] rtp.c: Oooh, 'SIP/201-08808278' was 0.0.0.0:0/(format 8) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Sending reinvite on SIP '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' - It's audio soon redirected to IP 192.168.1.208 [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Strict routing enforced for session 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 [Nov 16 11:45:09] VERBOSE[8476] logger.c: set_destination: Parsing for address/port to send to [Nov 16 11:45:09] VERBOSE[8476] logger.c: set_destination: set destination to 192.168.1.212, port 5060 [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 16 11:45:09] VERBOSE[8476] logger.c: Audio is at 192.168.1.156 port 15700 [Nov 16 11:45:09] VERBOSE[8476] logger.c: Adding codec 0x8 (alaw) to SDP [Nov 16 11:45:09] VERBOSE[8476] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: -- Done with adding codecs to SDP [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Initializing already initialized SIP dialog 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (presumably reinvite) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 0: INVITE sip:200@192.168.1.212:5060;transport=udp SIP/2.0 (55) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK738d6501;rport (64) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 2: From: ;tag=as25534657 (44) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 3: To: "200" ;tag=1163673819 (48) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 4: Contact: (32) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 5: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (55) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 10: Supported: replaces (19) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 13: Content-Length: 240 (19) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Header 14: (0) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: v=0 (3) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: o=root 3728 3729 IN IP4 192.168.1.208 (37) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: s=session (9) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: c=IN IP4 192.168.1.208 (22) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: m=audio 10014 RTP/AVP 8 101 (27) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: a=ptime:20 (10) [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: Line: a=sendrecv (10) [Nov 16 11:45:09] VERBOSE[8476] logger.c: Reliably Transmitting (no NAT) to 192.168.1.212:5060: INVITE sip:200@192.168.1.212:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK738d6501;rport From: ;tag=as25534657 To: "200" ;tag=1163673819 Contact: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 3728 3729 IN IP4 192.168.1.208 s=session c=IN IP4 192.168.1.208 t=0 0 m=audio 10014 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Nov 16 11:45:09] DEBUG[8476] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #929 [Nov 16 11:45:09] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.212:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK738d6501;rport=5060;received=192.168.1.156 User-Agent: Kapanga Softphone Desktop 1.00/2152 From: ;tag=as25534657 To: "200" ;tag=1163673819 Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 102 INVITE Contact: Supported: timer, replaces Content-Type: application/sdp Content-Length: 276 v=0 o=200 1163667398 1163673819 IN IP4 192.168.1.212 s=Kapanga [1163667398] c=IN IP4 192.168.1.212 t=0 0 m=audio 5144 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=sendrecv a=silenceSupp:off - - - a=maxptime:20 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK738d6501;rport=5060;received=192.168.1.156 (92) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 2: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 3: From: ;tag=as25534657 (44) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 4: To: "200" ;tag=1163673819 (48) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (55) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 7: Contact: (51) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 8: Supported: timer, replaces (26) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 9: Content-Type: application/sdp (29) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 10: Content-Length: 276 (19) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 11: (0) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: v=0 (3) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: o=200 1163667398 1163673819 IN IP4 192.168.1.212 (48) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: s=Kapanga [1163667398] (22) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: c=IN IP4 192.168.1.212 (22) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: m=audio 5144 RTP/AVP 8 101 (26) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: a=sendrecv (10) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: a=silenceSupp:off - - - (23) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: a=maxptime:20 (13) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: a=ptime:20 (10) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Nov 16 11:45:09] VERBOSE[3749] logger.c: --- (11 headers 13 lines) --- [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Acked pending invite 102 [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #929 [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Stopping retransmission on '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' of Request 102: Match Not Found [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: SIP response 200 to RE-invite on outgoing call 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 [Nov 16 11:45:09] VERBOSE[3749] logger.c: Found RTP audio format 8 [Nov 16 11:45:09] VERBOSE[3749] logger.c: Found RTP audio format 101 [Nov 16 11:45:09] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.212:5144 [Nov 16 11:45:09] VERBOSE[3749] logger.c: Found description format PCMA for ID 8 [Nov 16 11:45:09] VERBOSE[3749] logger.c: Found description format telephone-event for ID 101 [Nov 16 11:45:09] VERBOSE[3749] logger.c: Got unsupported a:fmtp in SDP offer [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: T38 state changed to 0 on channel SIP/200-088020c8 [Nov 16 11:45:09] VERBOSE[3749] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Nov 16 11:45:09] VERBOSE[3749] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 16 11:45:09] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.212:5144 [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: We have an owner, now see if we need to change this call [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Updating call counter for outgoing call [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: build_route: Contact hop: [Nov 16 11:45:09] VERBOSE[3749] logger.c: list_route: hop: [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Strange... The other side of the bridge does not have a udptl struct [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: T38 state changed to 0 on channel SIP [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: T38 state changed to 0 on channel SIP/200-088020c8 [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Strict routing enforced for session 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 [Nov 16 11:45:09] VERBOSE[3749] logger.c: set_destination: Parsing for address/port to send to [Nov 16 11:45:09] VERBOSE[3749] logger.c: set_destination: set destination to 192.168.1.212, port 5060 [Nov 16 11:45:09] VERBOSE[3749] logger.c: Transmitting (no NAT) to 192.168.1.212:5060: ACK sip:200@192.168.1.212:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK7bd46b1f;rport From: ;tag=as25534657 To: "200" ;tag=1163673819 Contact: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 16 11:45:09] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> REGISTER sip:192.168.1.156:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKc129fe81c Max-Forwards: 70 Content-Length: 0 To: 201 From: 201 ;tag=8f916ba381e7cf3 Call-ID: e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 CSeq: 474043496 REGISTER Contact: 201 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Authorization:Digest response="c8106c7ed078fa51d5df9ee2e1d7b997",username="201",realm="asterisk",nonce="0869101d",algorithm=MD5,uri="sip:192.168.1.156:5060" User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 <-------------> [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 0: REGISTER sip:192.168.1.156:5060 SIP/2.0 (39) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKc129fe81c (59) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 3: Content-Length: 0 (17) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 4: To: 201 (36) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 5: From: 201 ;tag=8f916ba381e7cf3 (58) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 6: Call-ID: e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 (55) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 7: CSeq: 474043496 REGISTER (24) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 8: Contact: 201 (41) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 9: Allow: NOTIFY (13) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 10: Allow: REFER (12) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 11: Allow: OPTIONS (14) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 12: Allow: INVITE (13) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 13: Allow: ACK (10) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 14: Allow: CANCEL (13) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 15: Allow: BYE (10) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 16: Authorization:Digest response="c8106c7ed078fa51d5df9ee2e1d7b997",username="201",realm="asterisk",nonce="0869101d",algorithm=MD5,uri="sip:192.168.1.156:5060" (156) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 17: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 18: (0) [Nov 16 11:45:09] VERBOSE[3749] logger.c: --- (18 headers 0 lines) --- [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Allocating new SIP dialog for e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 - REGISTER (No RTP) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 16 11:45:09] VERBOSE[3749] logger.c: Using latest REGISTER request as basis request [Nov 16 11:45:09] VERBOSE[3749] logger.c: Sending to 192.168.1.208 : 5060 (no NAT) [Nov 16 11:45:09] VERBOSE[3749] logger.c: <--- Transmitting (no NAT) to 192.168.1.208:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKc129fe81c;received=192.168.1.208 From: 201 ;tag=8f916ba381e7cf3 To: 201 Call-ID: e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 CSeq: 474043496 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Nov 16 11:45:09] VERBOSE[3749] logger.c: <--- Transmitting (no NAT) to 192.168.1.208:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKc129fe81c;received=192.168.1.208 From: 201 ;tag=8f916ba381e7cf3 To: 201 ;tag=as185e8602 Call-ID: e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 CSeq: 474043496 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7f1f72db" Content-Length: 0 <------------> [Nov 16 11:45:09] VERBOSE[3749] logger.c: Scheduling destruction of SIP dialog 'e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208' in 32000 ms (Method: REGISTER) [Nov 16 11:45:09] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> REGISTER sip:192.168.1.156:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKdcd9761be Max-Forwards: 70 Content-Length: 0 To: 201 From: 201 ;tag=8f916ba381e7cf3 Call-ID: e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 CSeq: 474043497 REGISTER Contact: 201 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Authorization:Digest response="08d992580a9f693c6f585756f9d6af28",username="201",realm="asterisk",nonce="7f1f72db",algorithm=MD5,uri="sip:192.168.1.156:5060" User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 <-------------> [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 0: REGISTER sip:192.168.1.156:5060 SIP/2.0 (39) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKdcd9761be (59) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 3: Content-Length: 0 (17) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 4: To: 201 (36) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 5: From: 201 ;tag=8f916ba381e7cf3 (58) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 6: Call-ID: e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 (55) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 7: CSeq: 474043497 REGISTER (24) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 8: Contact: 201 (41) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 9: Allow: NOTIFY (13) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 10: Allow: REFER (12) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 11: Allow: OPTIONS (14) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 12: Allow: INVITE (13) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 13: Allow: ACK (10) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 14: Allow: CANCEL (13) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 15: Allow: BYE (10) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 16: Authorization:Digest response="08d992580a9f693c6f585756f9d6af28",username="201",realm="asterisk",nonce="7f1f72db",algorithm=MD5,uri="sip:192.168.1.156:5060" (156) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 17: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: Header 18: (0) [Nov 16 11:45:09] VERBOSE[3749] logger.c: --- (18 headers 0 lines) --- [Nov 16 11:45:09] DEBUG[3749] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Nov 16 11:45:09] VERBOSE[3749] logger.c: Using latest REGISTER request as basis request [Nov 16 11:45:09] VERBOSE[3749] logger.c: Sending to 192.168.1.208 : 5060 (no NAT) [Nov 16 11:45:09] VERBOSE[3749] logger.c: <--- Transmitting (no NAT) to 192.168.1.208:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKdcd9761be;received=192.168.1.208 From: 201 ;tag=8f916ba381e7cf3 To: 201 Call-ID: e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 CSeq: 474043497 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Nov 16 11:45:09] VERBOSE[3749] logger.c: <--- Transmitting (no NAT) to 192.168.1.208:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKdcd9761be;received=192.168.1.208 From: 201 ;tag=8f916ba381e7cf3 To: 201 ;tag=as185e8602 Call-ID: e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 CSeq: 474043497 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Thu, 16 Nov 2006 10:45:09 GMT Content-Length: 0 <------------> [Nov 16 11:45:09] DEBUG[3749] devicestate.c: Notification of state change to be queued on device/channel SIP/201 [Nov 16 11:45:09] VERBOSE[3749] logger.c: Scheduling destruction of SIP dialog 'e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208' in 32000 ms (Method: REGISTER) [Nov 16 11:45:09] DEBUG[3731] devicestate.c: No provider found, checking channel drivers for SIP - 201 [Nov 16 11:45:09] DEBUG[3731] chan_sip.c: Checking device state for peer 201 [Nov 16 11:45:09] DEBUG[3731] devicestate.c: Changing state for SIP/201 - state 1 (Not in use) [Nov 16 11:45:09] DEBUG[8495] app_queue.c: Device 'SIP/201' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 16 11:45:10] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> BYE sip:200@192.168.1.156 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKe8f6d9a83 Max-Forwards: 70 Content-Length: 0 To: "200" ;tag=as2e312935 From: ;tag=86878122aefef67 Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 31662864 BYE Supported: timer Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Supported: replaces User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 <-------------> [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 0: BYE sip:200@192.168.1.156 SIP/2.0 (33) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKe8f6d9a83 (59) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 2: Max-Forwards: 70 (16) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 3: Content-Length: 0 (17) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 4: To: "200" ;tag=as2e312935 (48) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 5: From: ;tag=86878122aefef67 (54) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 6: Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 (55) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 7: CSeq: 31662864 BYE (18) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 8: Supported: timer (16) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 9: Allow: NOTIFY (13) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 10: Allow: REFER (12) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 11: Allow: OPTIONS (14) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 12: Allow: INVITE (13) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 13: Allow: ACK (10) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 14: Allow: CANCEL (13) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 15: Allow: BYE (10) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 16: Supported: replaces (19) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 17: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 18: (0) [Nov 16 11:45:10] VERBOSE[3749] logger.c: --- (18 headers 0 lines) --- [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Nov 16 11:45:10] VERBOSE[3749] logger.c: Sending to 192.168.1.208 : 5060 (no NAT) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Received bye, issuing owner hangup .[Nov 16 11:45:10] VERBOSE[3749] logger.c: <--- Transmitting (no NAT) to 192.168.1.208:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.208:5060;branch=z9hG4bKe8f6d9a83;received=192.168.1.208 From: ;tag=86878122aefef67 To: "200" ;tag=as2e312935 Call-ID: 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156 CSeq: 31662864 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Nov 16 11:45:10] DEBUG[8476] rtp.c: Oooh, got a hangup [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Sending reinvite on SIP '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' - It's audio soon redirected to IP 192.168.1.156 [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Strict routing enforced for session 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 [Nov 16 11:45:10] VERBOSE[8476] logger.c: set_destination: Parsing for address/port to send to [Nov 16 11:45:10] VERBOSE[8476] logger.c: set_destination: set destination to 192.168.1.212, port 5060 [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Nov 16 11:45:10] VERBOSE[8476] logger.c: Audio is at 192.168.1.156 port 15700 [Nov 16 11:45:10] VERBOSE[8476] logger.c: Adding codec 0x8 (alaw) to SDP [Nov 16 11:45:10] VERBOSE[8476] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: -- Done with adding codecs to SDP [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Initializing already initialized SIP dialog 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (presumably reinvite) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 0: INVITE sip:200@192.168.1.212:5060;transport=udp SIP/2.0 (55) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK33916900;rport (64) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 2: From: ;tag=as25534657 (44) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 3: To: "200" ;tag=1163673819 (48) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 4: Contact: (32) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 5: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (55) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 6: CSeq: 103 INVITE (16) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 10: Supported: replaces (19) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 11: X-asterisk-Info: SIP re-invite (External RTP bridge) (52) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 13: Content-Length: 240 (19) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Header 14: (0) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: v=0 (3) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: o=root 3728 3730 IN IP4 192.168.1.156 (37) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: s=session (9) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: c=IN IP4 192.168.1.156 (22) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: m=audio 15700 RTP/AVP 8 101 (27) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: a=ptime:20 (10) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Line: a=sendrecv (10) [Nov 16 11:45:10] VERBOSE[8476] logger.c: Reliably Transmitting (no NAT) to 192.168.1.212:5060: INVITE sip:200@192.168.1.212:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK33916900;rport From: ;tag=as25534657 To: "200" ;tag=1163673819 Contact: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 3728 3730 IN IP4 192.168.1.156 s=session c=IN IP4 192.168.1.156 t=0 0 m=audio 15700 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #933 [Nov 16 11:45:10] DEBUG[8476] channel.c: Returning from native bridge, channels: SIP/200-088020c8, SIP/201-08808278 [Nov 16 11:45:10] DEBUG[8476] channel.c: Hanging up channel 'SIP/201-08808278' [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Hangup call SIP/201-08808278, SIP callid 7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: update_call_counter(201) - decrement call limit counter on hangup [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Updating call counter for incoming call [Nov 16 11:45:10] DEBUG[8476] devicestate.c: Notification of state change to be queued on device/channel SIP/201-08808278 [Nov 16 11:45:10] DEBUG[8476] rtp.c: Channel '' has no RTP, not doing anything [Nov 16 11:45:10] DEBUG[8476] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Nov 16 11:45:10] DEBUG[8476] pbx.c: Spawn extension (from-sip,201,1) exited non-zero on 'SIP/200-088020c8' [Nov 16 11:45:10] VERBOSE[8476] logger.c: == Spawn extension (from-sip, 201, 1) exited non-zero on 'SIP/200-088020c8' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '"200" <200>' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '200' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '201' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is 'from-sip' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is 'SIP/200-088020c8' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is 'SIP/201-08808278' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is 'Dial' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is 'SIP/201|300' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '2006-11-16 11:44:20' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '2006-11-16 11:44:29' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '2006-11-16 11:45:10' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '50' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '41' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is 'ANSWERED' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is 'DOCUMENTATION' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '1163673860.22' [Nov 16 11:45:10] DEBUG[8476] pbx.c: Function result is '' [Nov 16 11:45:10] DEBUG[8476] channel.c: Hanging up channel 'SIP/200-088020c8' [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Hangup call SIP/200-088020c8, SIP callid 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156) [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: update_call_counter(200) - decrement call limit counter on hangup [Nov 16 11:45:10] DEBUG[8476] chan_sip.c: Updating call counter for outgoing call [Nov 16 11:45:10] VERBOSE[8476] logger.c: Scheduling destruction of SIP dialog '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' in 32000 ms (Method: ACK) [Nov 16 11:45:10] DEBUG[8476] devicestate.c: Notification of state change to be queued on device/channel SIP/200-088020c8 [Nov 16 11:45:10] DEBUG[3731] devicestate.c: No provider found, checking channel drivers for SIP - 201 [Nov 16 11:45:10] DEBUG[3731] chan_sip.c: Checking device state for peer 201 [Nov 16 11:45:10] DEBUG[3731] devicestate.c: Changing state for SIP/201 - state 1 (Not in use) [Nov 16 11:45:10] DEBUG[3731] devicestate.c: No provider found, checking channel drivers for SIP - 200 [Nov 16 11:45:10] DEBUG[3731] chan_sip.c: Checking device state for peer 200 [Nov 16 11:45:10] DEBUG[3731] devicestate.c: Changing state for SIP/200 - state 1 (Not in use) [Nov 16 11:45:10] DEBUG[8496] app_queue.c: Device 'SIP/201' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 16 11:45:10] DEBUG[8497] app_queue.c: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Nov 16 11:45:10] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.212:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK33916900;rport=5060;received=192.168.1.156 User-Agent: Kapanga Softphone Desktop 1.00/2152 From: ;tag=as25534657 To: "200" ;tag=1163673819 Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 103 INVITE Contact: Supported: timer, replaces Content-Type: application/sdp Content-Length: 276 v=0 o=200 1163667398 1163673819 IN IP4 192.168.1.212 s=Kapanga [1163667398] c=IN IP4 192.168.1.212 t=0 0 m=audio 5144 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=sendrecv a=silenceSupp:off - - - a=maxptime:20 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK33916900;rport=5060;received=192.168.1.156 (92) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 2: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 3: From: ;tag=as25534657 (44) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 4: To: "200" ;tag=1163673819 (48) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (55) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 6: CSeq: 103 INVITE (16) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 7: Contact: (51) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 8: Supported: timer, replaces (26) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 9: Content-Type: application/sdp (29) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 10: Content-Length: 276 (19) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 11: (0) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: v=0 (3) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: o=200 1163667398 1163673819 IN IP4 192.168.1.212 (48) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: s=Kapanga [1163667398] (22) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: c=IN IP4 192.168.1.212 (22) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: t=0 0 (5) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: m=audio 5144 RTP/AVP 8 101 (26) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: a=sendrecv (10) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: a=silenceSupp:off - - - (23) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: a=maxptime:20 (13) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: a=ptime:20 (10) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Line: a=fmtp:101 0-16 (15) [Nov 16 11:45:10] VERBOSE[3749] logger.c: --- (11 headers 13 lines) --- [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Acked pending invite 103 [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #933 [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Stopping retransmission on '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' of Request 103: Match Not Found [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: SIP response 200 to standard invite [Nov 16 11:45:10] VERBOSE[3749] logger.c: Found RTP audio format 8 [Nov 16 11:45:10] VERBOSE[3749] logger.c: Found RTP audio format 101 [Nov 16 11:45:10] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.212:5144 [Nov 16 11:45:10] VERBOSE[3749] logger.c: Found description format PCMA for ID 8 [Nov 16 11:45:10] VERBOSE[3749] logger.c: Found description format telephone-event for ID 101 [Nov 16 11:45:10] VERBOSE[3749] logger.c: Got unsupported a:fmtp in SDP offer [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: T38 state changed to 0 on channel [Nov 16 11:45:10] VERBOSE[3749] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Nov 16 11:45:10] VERBOSE[3749] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 16 11:45:10] VERBOSE[3749] logger.c: Peer audio RTP is at port 192.168.1.212:5144 [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Updating call counter for outgoing call [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: build_route: Retaining previous route: [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Strict routing enforced for session 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 [Nov 16 11:45:10] VERBOSE[3749] logger.c: set_destination: Parsing for address/port to send to [Nov 16 11:45:10] VERBOSE[3749] logger.c: set_destination: set destination to 192.168.1.212, port 5060 [Nov 16 11:45:10] VERBOSE[3749] logger.c: Transmitting (no NAT) to 192.168.1.212:5060: ACK sip:200@192.168.1.212:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK2497d466;rport From: ;tag=as25534657 To: "200" ;tag=1163673819 Contact: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Strict routing enforced for session 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 [Nov 16 11:45:10] VERBOSE[3749] logger.c: set_destination: Parsing for address/port to send to [Nov 16 11:45:10] VERBOSE[3749] logger.c: set_destination: set destination to 192.168.1.212, port 5060 [Nov 16 11:45:10] VERBOSE[3749] logger.c: Reliably Transmitting (no NAT) to 192.168.1.212:5060: BYE sip:200@192.168.1.212:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK7896351a;rport From: ;tag=as25534657 To: "200" ;tag=1163673819 Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #935 [Nov 16 11:45:10] VERBOSE[3749] logger.c: Scheduling destruction of SIP dialog '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' in 32000 ms (Method: ACK) [Nov 16 11:45:10] VERBOSE[3749] logger.c: Really destroying SIP dialog '7144f1d80f30cc197af9ca7a31b3daa4@192.168.1.156' Method: BYE [Nov 16 11:45:10] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.212:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK7896351a;rport=5060;received=192.168.1.156 User-Agent: Kapanga Softphone Desktop 1.00/2152 From: ;tag=as25534657 To: "200" ;tag=1163673819 Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 CSeq: 104 BYE Contact: Supported: timer, replaces Content-Length: 0 <-------------> [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK7896351a;rport=5060;received=192.168.1.156 (92) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 2: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 3: From: ;tag=as25534657 (44) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 4: To: "200" ;tag=1163673819 (48) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 7592B50EFB869505BB93F17A6E77BE65@192.168.1.156 (55) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 6: CSeq: 104 BYE (13) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 7: Contact: (51) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 8: Supported: timer, replaces (26) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 9: Content-Length: 0 (17) [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Header 10: (0) [Nov 16 11:45:10] VERBOSE[3749] logger.c: --- (10 headers 0 lines) --- [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #935 [Nov 16 11:45:10] DEBUG[3749] chan_sip.c: Stopping retransmission on '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' of Request 104: Match Not Found [Nov 16 11:45:10] VERBOSE[3749] logger.c: Really destroying SIP dialog '7592B50EFB869505BB93F17A6E77BE65@192.168.1.156' Method: ACK [Nov 16 11:45:29] DEBUG[3749] chan_sip.c: Auto destroying SIP dialog '7479F055307F434C30B8470A943785A3@192.168.1.156' [Nov 16 11:45:29] DEBUG[3749] chan_sip.c: Destroying SIP dialog 7479F055307F434C30B8470A943785A3@192.168.1.156 [Nov 16 11:45:29] VERBOSE[3749] logger.c: Really destroying SIP dialog '7479F055307F434C30B8470A943785A3@192.168.1.156' Method: OPTIONS [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 0: OPTIONS sip:201@192.168.1.208:5060 SIP/2.0 (42) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK0a78bc72;rport (64) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 2: From: "asterisk" ;tag=as527acdd4 (60) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 3: To: (32) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 4: Contact: (37) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 77f56f7663bfc8e26c28d8ff1435ae42@192.168.1.156 (55) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 9: Date: Thu, 16 Nov 2006 10:45:32 GMT (35) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 11: Supported: replaces (19) [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: Header 12: Content-Length: 0 (17) [Nov 16 11:45:32] VERBOSE[3749] logger.c: Reliably Transmitting (no NAT) to 192.168.1.208:5060: OPTIONS sip:201@192.168.1.208:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK0a78bc72;rport From: "asterisk" ;tag=as527acdd4 To: Contact: Call-ID: 77f56f7663bfc8e26c28d8ff1435ae42@192.168.1.156 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 16 Nov 2006 10:45:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Nov 16 11:45:32] DEBUG[3749] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #937 [Nov 16 11:45:33] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.208:5060 ---> SIP/2.0 200 OK Call-ID: 77f56f7663bfc8e26c28d8ff1435ae42@192.168.1.156 CSeq: 102 OPTIONS From: "asterisk" ;tag=as527acdd4 To: ;tag=dd6f48730248fa7 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK0a78bc72;rport Content-Length: 0 Allow: NOTIFY Allow: REFER Allow: OPTIONS Allow: INVITE Allow: ACK Allow: CANCEL Allow: BYE Contact: Supported: replaces User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 <-------------> [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 1: Call-ID: 77f56f7663bfc8e26c28d8ff1435ae42@192.168.1.156 (55) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 2: CSeq: 102 OPTIONS (17) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 3: From: "asterisk" ;tag=as527acdd4 (60) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 4: To: ;tag=dd6f48730248fa7 (52) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 5: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK0a78bc72;rport (64) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 6: Content-Length: 0 (17) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 7: Allow: NOTIFY (13) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 8: Allow: REFER (12) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 9: Allow: OPTIONS (14) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 10: Allow: INVITE (13) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 11: Allow: ACK (10) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 12: Allow: CANCEL (13) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 13: Allow: BYE (10) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 14: Contact: (37) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 15: Supported: replaces (19) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 16: User-Agent: Brcm Callctrl/1.5.1.1 MxSF/v3.2.6.26 (48) [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Header 17: (0) [Nov 16 11:45:33] VERBOSE[3749] logger.c: --- (17 headers 0 lines) --- [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #937 [Nov 16 11:45:33] DEBUG[3749] chan_sip.c: Stopping retransmission on '77f56f7663bfc8e26c28d8ff1435ae42@192.168.1.156' of Request 102: Match Not Found [Nov 16 11:45:33] VERBOSE[3749] logger.c: Really destroying SIP dialog '77f56f7663bfc8e26c28d8ff1435ae42@192.168.1.156' Method: OPTIONS [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 0: OPTIONS sip:200@192.168.1.212:5060;transport=udp SIP/2.0 (56) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK4294757e;rport (64) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 2: From: "asterisk" ;tag=as08366598 (60) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 3: To: (46) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 4: Contact: (37) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 2974a34a065a39a15a95aff63dbfac07@192.168.1.156 (55) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 9: Date: Thu, 16 Nov 2006 10:45:37 GMT (35) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 11: Supported: replaces (19) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 12: Content-Length: 0 (17) [Nov 16 11:45:37] VERBOSE[3749] logger.c: Reliably Transmitting (no NAT) to 192.168.1.212:5060: OPTIONS sip:200@192.168.1.212:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK4294757e;rport From: "asterisk" ;tag=as08366598 To: Contact: Call-ID: 2974a34a065a39a15a95aff63dbfac07@192.168.1.156 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 16 Nov 2006 10:45:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #940 [Nov 16 11:45:37] VERBOSE[3749] logger.c: <--- SIP read from 192.168.1.212:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK4294757e;rport=5060;received=192.168.1.156 User-Agent: Kapanga Softphone Desktop 1.00/2152 From: ;tag=1163673896 To: "asterisk" ;tag=as08366598 Call-ID: 2974a34a065a39a15a95aff63dbfac07@192.168.1.156 CSeq: 102 OPTIONS Contact: Supported: timer, replaces Content-Length: 0 <-------------> [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 0: SIP/2.0 200 OK (14) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 1: Via: SIP/2.0/UDP 192.168.1.156:5060;branch=z9hG4bK4294757e;rport=5060;received=192.168.1.156 (92) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 2: User-Agent: Kapanga Softphone Desktop 1.00/2152 (47) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 3: From: ;tag=1163673896 (63) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 4: To: "asterisk" ;tag=as08366598 (58) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 5: Call-ID: 2974a34a065a39a15a95aff63dbfac07@192.168.1.156 (55) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 7: Contact: (51) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 8: Supported: timer, replaces (26) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 9: Content-Length: 0 (17) [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Header 10: (0) [Nov 16 11:45:37] VERBOSE[3749] logger.c: --- (10 headers 0 lines) --- [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #940 [Nov 16 11:45:37] DEBUG[3749] chan_sip.c: Stopping retransmission on '2974a34a065a39a15a95aff63dbfac07@192.168.1.156' of Request 102: Match Not Found [Nov 16 11:45:37] VERBOSE[3749] logger.c: Really destroying SIP dialog '2974a34a065a39a15a95aff63dbfac07@192.168.1.156' Method: OPTIONS [Nov 16 11:45:41] DEBUG[3749] chan_sip.c: Auto destroying SIP dialog 'e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208' [Nov 16 11:45:41] DEBUG[3749] chan_sip.c: Destroying SIP dialog e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208 [Nov 16 11:45:41] VERBOSE[3749] logger.c: Really destroying SIP dialog 'e45b4c37bfb2e1ed7cfebcf5f0129304@192.168.1.208' Method: REGISTER [Nov 16 11:45:58] VERBOSE[3749] logger.c: