Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 0: INVITE sip:085661098@10.0.5.15:5060;user=phone SIP/2.0 (54) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.129;branch=z9hG4bKaab88cc763797956 (58) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 2: From: "Line1" ;tag=177D7E89-F26F3C02 (62) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 3: To: (40) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 4: CSeq: 1 INVITE (14) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 60f08dfd-b4a60c93-550fc47c@10.0.5.129 (46) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 9: Supported: 100rel,replaces (26) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 11: Max-Forwards: 70 (16) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 12: Content-Type: application/sdp (29) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 13: Content-Length: 233 (19) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 14: (0) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: o=- 1159700477 1159700477 IN IP4 10.0.5.129 (43) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: s=Polycom IP Phone (18) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: c=IN IP4 10.0.5.129 (19) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: m=audio 2236 RTP/AVP 8 0 18 101 (31) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for 60f08dfd-b4a60c93-550fc47c@10.0.5.129 - INVITE (With RTP) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Begin: parsing SIP "Supported: 100rel,replaces" Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Found SIP option: -100rel- Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Matched SIP option: 100rel Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Found SIP option: -replaces- Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Matched SIP option: replaces Oct 1 21:01:21 DEBUG[10546] chan_sip.c: * SIP extension value: 3 for call 60f08dfd-b4a60c93-550fc47c@10.0.5.129 Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Setting NAT on RTP to 0 Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Checking SIP call limits for device polycom_p1 Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Updating call counter for incoming call Oct 1 21:01:21 DEBUG[10546] chan_sip.c: build_route: Contact hop: Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.129;branch=z9hG4bKaab88cc763797956;received=10.0.5.129 (78) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 2: From: "Line1" ;tag=177D7E89-F26F3C02 (62) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 3: To: (40) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 4: Call-ID: 60f08dfd-b4a60c93-550fc47c@10.0.5.129 (46) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 5: CSeq: 1 INVITE (14) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 8: Contact: (34) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:21 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:21 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_p1 Oct 1 21:01:21 DEBUG[10538] channel.c: Avoiding initial deadlock for 'SIP/polycom_p1-081fc808' Oct 1 21:01:21 DEBUG[10562] pbx.c: Launching 'Macro' Oct 1 21:01:21 VERBOSE[10562] logger.c: -- Executing Macro("SIP/polycom_p1-081fc808", "dialout|145685661098") in new stack Oct 1 21:01:21 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_p1 - state 2 (In use) Oct 1 21:01:21 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_p1 Oct 1 21:01:21 DEBUG[10562] pbx.c: Function result is 'SIP/polycom_p1' Oct 1 21:01:21 DEBUG[10562] pbx.c: Launching 'Set' Oct 1 21:01:21 VERBOSE[10562] logger.c: -- Executing Set("SIP/polycom_p1-081fc808", "member=SIP/polycom_p1") in new stack Oct 1 21:01:21 DEBUG[10562] pbx.c: Launching 'NoOp' Oct 1 21:01:21 VERBOSE[10562] logger.c: -- Executing NoOp("SIP/polycom_p1-081fc808", "") in new stack Oct 1 21:01:21 DEBUG[10562] pbx.c: Launching 'NoOp' Oct 1 21:01:21 VERBOSE[10562] logger.c: -- Executing NoOp("SIP/polycom_p1-081fc808", "") in new stack Oct 1 21:01:21 DEBUG[10562] pbx.c: Launching 'SetCallerID' Oct 1 21:01:21 VERBOSE[10562] logger.c: -- Executing SetCallerID("SIP/polycom_p1-081fc808", "85661000") in new stack Oct 1 21:01:21 DEBUG[10562] pbx.c: Launching 'Dial' Oct 1 21:01:21 VERBOSE[10562] logger.c: -- Executing Dial("SIP/polycom_p1-081fc808", "Zap/G2/145685661098") in new stack Oct 1 21:01:21 DEBUG[10562] chan_zap.c: Using channel 10 Oct 1 21:01:21 DEBUG[10538] devicestate.c: Changing state for Zap/10 - state 2 (In use) Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable STACK-macro-dialout-s-5. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable MACRO_DEPTH. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable STACK-macro-dialout-s-4. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable STACK-macro-dialout-s-3. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable STACK-macro-dialout-s-2. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable member. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable STACK-macro-dialout-s-1. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable ARG1. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable MACRO_PRIORITY. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable MACRO_CONTEXT. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable MACRO_EXTEN. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable STACK-inside-085661098-1. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable SIPCALLID. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable SIPUSERAGENT. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable SIPDOMAIN. Oct 1 21:01:21 DEBUG[10562] channel.c: Not copying variable SIPURI. Oct 1 21:01:21 VERBOSE[10562] logger.c: -- Requested transfer capability: 0x00 - SPEECH Oct 1 21:01:21 DEBUG[10538] channel.c: Avoiding initial deadlock for 'Zap/10-1' Oct 1 21:01:21 DEBUG[10538] channel.c: Avoiding initial deadlock for 'Zap/10-1' Oct 1 21:01:21 VERBOSE[10562] logger.c: -- Called G2/145685661098 Oct 1 21:01:21 DEBUG[10538] devicestate.c: Changing state for Zap/10 - state 2 (In use) Oct 1 21:01:21 DEBUG[10563] app_queue.c: Device 'SIP/polycom_p1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:21 DEBUG[10564] app_queue.c: Device 'Zap/10' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:21 DEBUG[10565] app_queue.c: Device 'Zap/10' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 0: SIP/2.0 183 Session Progress (28) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.129;branch=z9hG4bKaab88cc763797956;received=10.0.5.129 (78) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 2: From: "Line1" ;tag=177D7E89-F26F3C02 (62) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 3: To: ;tag=as3041f41c (55) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 4: Call-ID: 60f08dfd-b4a60c93-550fc47c@10.0.5.129 (46) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 5: CSeq: 1 INVITE (14) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 8: Contact: (34) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 10: Content-Length: 210 (19) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Header 11: (0) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: v=0 (3) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: o=root 10562 10562 IN IP4 10.0.5.15 (35) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: s=session (9) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: m=audio 10304 RTP/AVP 8 101 (27) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:21 DEBUG[10562] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:21 DEBUG[10562] rtp.c: Ooh, format changed from unknown to alaw Oct 1 21:01:21 DEBUG[10543] chan_zap.c: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/10 span 1 Oct 1 21:01:21 VERBOSE[10562] logger.c: -- Zap/10-1 is proceeding passing it to SIP/polycom_p1-081fc808 Oct 1 21:01:22 DEBUG[10538] devicestate.c: Changing state for Zap/1 - state 2 (In use) Oct 1 21:01:22 VERBOSE[10543] logger.c: -- Accepting call from '0285661000' to '85661098' on channel 0/1, span 1 Oct 1 21:01:22 DEBUG[10543] chan_zap.c: Enabled echo cancellation on channel 1 Oct 1 21:01:22 DEBUG[10566] pbx.c: Launching 'Goto' Oct 1 21:01:22 VERBOSE[10566] logger.c: -- Executing Goto("Zap/1-1", "open|s|1") in new stack Oct 1 21:01:22 VERBOSE[10566] logger.c: -- Goto (open,s,1) Oct 1 21:01:22 DEBUG[10566] pbx.c: Launching 'Answer' Oct 1 21:01:22 VERBOSE[10566] logger.c: -- Executing Answer("Zap/1-1", "") in new stack Oct 1 21:01:22 DEBUG[10538] channel.c: Avoiding initial deadlock for 'Zap/1-1' Oct 1 21:01:22 DEBUG[10566] pbx.c: Launching 'Ringing' Oct 1 21:01:22 VERBOSE[10566] logger.c: -- Executing Ringing("Zap/1-1", "") in new stack Oct 1 21:01:22 DEBUG[10538] devicestate.c: Changing state for Zap/1 - state 2 (In use) Oct 1 21:01:22 DEBUG[10566] chan_zap.c: Requested indication 3 on channel Zap/1-1 Oct 1 21:01:22 DEBUG[10566] pbx.c: Launching 'Wait' Oct 1 21:01:22 VERBOSE[10566] logger.c: -- Executing Wait("Zap/1-1", "3") in new stack Oct 1 21:01:22 DEBUG[10567] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:22 DEBUG[10568] app_queue.c: Device 'Zap/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:22 DEBUG[10543] chan_zap.c: Enabled echo cancellation on channel 10 Oct 1 21:01:22 VERBOSE[10562] logger.c: -- Zap/10-1 answered SIP/polycom_p1-081fc808 Oct 1 21:01:22 DEBUG[10562] chan_sip.c: sip_answer(SIP/polycom_p1-081fc808) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.129;branch=z9hG4bKaab88cc763797956;received=10.0.5.129 (78) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 2: From: "Line1" ;tag=177D7E89-F26F3C02 (62) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 3: To: ;tag=as3041f41c (55) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 4: Call-ID: 60f08dfd-b4a60c93-550fc47c@10.0.5.129 (46) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 5: CSeq: 1 INVITE (14) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 8: Contact: (34) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 10: Content-Length: 210 (19) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Header 11: (0) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: v=0 (3) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: o=root 10562 10563 IN IP4 10.0.5.15 (35) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: s=session (9) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: m=audio 10304 RTP/AVP 8 101 (27) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:22 DEBUG[10562] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #113 Oct 1 21:01:22 DEBUG[10538] devicestate.c: Changing state for Zap/10 - state 2 (In use) Oct 1 21:01:22 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_p1 Oct 1 21:01:22 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_p1 - state 2 (In use) Oct 1 21:01:22 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_p1 Oct 1 21:01:22 DEBUG[10569] app_queue.c: Device 'Zap/10' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:22 DEBUG[10570] app_queue.c: Device 'SIP/polycom_p1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 0: ACK sip:085661098@10.0.5.15 SIP/2.0 (35) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.129;branch=z9hG4bKfb8cb9dfAACBA82E (58) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 2: From: "Line1" ;tag=177D7E89-F26F3C02 (62) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=as3041f41c (55) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 4: CSeq: 1 ACK (11) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 60f08dfd-b4a60c93-550fc47c@10.0.5.129 (46) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 9: Max-Forwards: 70 (16) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 0 (17) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:22 DEBUG[10546] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Oct 1 21:01:22 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #113 Oct 1 21:01:22 DEBUG[10546] chan_sip.c: Stopping retransmission on '60f08dfd-b4a60c93-550fc47c@10.0.5.129' of Response 1: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_l1@10.0.5.137 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0d718608;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as0e85e97c (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 31eb698b4e1ff74552d4f5ae4d95dc6a@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_l1@10.0.5.137 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0d718608;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as0e85e97c (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 31eb698b4e1ff74552d4f5ae4d95dc6a@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #114 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_q1@10.0.5.162 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK180865e9;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as5087446c (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 3f99051a14e8bdd71888193f698e05b5@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_q1@10.0.5.162 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK180865e9;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as5087446c (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 3f99051a14e8bdd71888193f698e05b5@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #116 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_c1@10.0.5.142 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK1ec99e37;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as7ae777db (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 0a43a6cf7c57fced6512b8f47cc3607b@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_c1@10.0.5.142 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK1ec99e37;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as7ae777db (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 0a43a6cf7c57fced6512b8f47cc3607b@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #118 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_b1@10.0.5.143 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0db5df53;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as4094a1ab (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 20762e2440cbf0fd62a60c21217d22ad@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_b1@10.0.5.143 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0db5df53;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as4094a1ab (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 20762e2440cbf0fd62a60c21217d22ad@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #120 Oct 1 21:01:25 DEBUG[10545] chan_iax2.c: Peer lastms 4, historicms 4, maxms 2000 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_a1@10.0.5.132 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK11d193b7;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as2e7c402d (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 2461564254f27dff64ec249257d788a8@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_a1@10.0.5.132 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK11d193b7;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as2e7c402d (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 2461564254f27dff64ec249257d788a8@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #122 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_i1@10.0.5.133 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK12a9ff9c;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as62cfdeb0 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 2925a64159679a5d6e037db47c896730@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 VERBOSE[10546] logger.c: 12 headers, 0 lines Oct 1 21:01:25 VERBOSE[10546] logger.c: Reliably Transmitting (no NAT) to 10.0.5.133:5060: OPTIONS sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK12a9ff9c;rport From: "asterisk" ;tag=as62cfdeb0 To: Contact: Call-ID: 2925a64159679a5d6e037db47c896730@10.0.5.15 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 01 Oct 2006 11:01:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_i1@10.0.5.133 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK12a9ff9c;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as62cfdeb0 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 2925a64159679a5d6e037db47c896730@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #124 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_h1@10.0.5.141 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK196fc6e8;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as0638f944 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 04fe6a451ee8cfa34de303711e6a47dd@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_h1@10.0.5.141 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK196fc6e8;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as0638f944 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 04fe6a451ee8cfa34de303711e6a47dd@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #126 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_f1@10.0.5.134 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4b45e57e;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as4fcef59b (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 7bc3c69b32b61cbf309192862b790a62@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_f1@10.0.5.134 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4b45e57e;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as4fcef59b (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 7bc3c69b32b61cbf309192862b790a62@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #128 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_g1@10.0.5.135 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK134d015e;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as74f64bca (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 1b8b09667e9df557542efb4f14c39cfa@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_g1@10.0.5.135 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK134d015e;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as74f64bca (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 1b8b09667e9df557542efb4f14c39cfa@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #130 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_j1@10.0.5.130 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK458aba33;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as49765fa6 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 44a157f51cc632a07fcc834d2e48c37a@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_j1@10.0.5.130 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK458aba33;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as49765fa6 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 44a157f51cc632a07fcc834d2e48c37a@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #132 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_m1@10.0.5.138 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK10cab6bc;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as2002f119 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 266b37c01942fadb6f6119a35230f854@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_m1@10.0.5.138 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK10cab6bc;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as2002f119 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 266b37c01942fadb6f6119a35230f854@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #134 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_n1@10.0.5.140 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK73b15e95;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as7beb00fc (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 0f3d488f53f529e3087462410ead5b86@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_n1@10.0.5.140 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK73b15e95;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as7beb00fc (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 0f3d488f53f529e3087462410ead5b86@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #136 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_o1@10.0.5.155 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK087ae2c2;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as6e3ccbf8 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 35d9df0a62eaddc10cd8f6b85ca7ec53@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_o1@10.0.5.155 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK087ae2c2;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as6e3ccbf8 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 35d9df0a62eaddc10cd8f6b85ca7ec53@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #138 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_k1@10.0.5.139 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK5bb5f849;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as6046627f (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 7d00a3a75faca130151daa8d0a13f657@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_k1@10.0.5.139 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK5bb5f849;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as6046627f (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 7d00a3a75faca130151daa8d0a13f657@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #140 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_p1@10.0.5.129 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6c229729;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as343f2592 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 562265857c5e6d886df63907376c98ad@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_p1@10.0.5.129 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6c229729;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as343f2592 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 562265857c5e6d886df63907376c98ad@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #142 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_e1@10.0.5.156 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6723bd76;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as56d90401 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 097a0aa53d977bf16122c0500125b16a@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: OPTIONS sip:polycom_e1@10.0.5.156 SIP/2.0 (41) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6723bd76;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as56d90401 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: (31) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: Contact: (33) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 097a0aa53d977bf16122c0500125b16a@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:25 GMT (35) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #144 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0d718608;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as0e85e97c (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=480A7332-D9D4BC1F (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 31eb698b4e1ff74552d4f5ae4d95dc6a@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #114 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '31eb698b4e1ff74552d4f5ae4d95dc6a@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0db5df53;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as4094a1ab (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=F6344315-F837464 (52) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 20762e2440cbf0fd62a60c21217d22ad@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #120 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '20762e2440cbf0fd62a60c21217d22ad@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK1ec99e37;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as7ae777db (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=A2DEF121-CDB2C30 (52) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 0a43a6cf7c57fced6512b8f47cc3607b@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #118 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '0a43a6cf7c57fced6512b8f47cc3607b@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK180865e9;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as5087446c (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=C91CA939-7836C0C0 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 3f99051a14e8bdd71888193f698e05b5@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #116 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '3f99051a14e8bdd71888193f698e05b5@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK11d193b7;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as2e7c402d (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=81CBBBCF-118EE180 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 2461564254f27dff64ec249257d788a8@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #122 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '2461564254f27dff64ec249257d788a8@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK12a9ff9c;rport From: "asterisk" ;tag=as62cfdeb0 To: ;tag=52A41FA8-6FEC7BF3 CSeq: 102 OPTIONS Call-ID: 2925a64159679a5d6e037db47c896730@10.0.5.15 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Content-Length: 0 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK12a9ff9c;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as62cfdeb0 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=52A41FA8-6FEC7BF3 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 2925a64159679a5d6e037db47c896730@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 VERBOSE[10546] logger.c: --- (10 headers 0 lines) --- Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #124 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '2925a64159679a5d6e037db47c896730@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 VERBOSE[10546] logger.c: Destroying call '2925a64159679a5d6e037db47c896730@10.0.5.15' Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK134d015e;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as74f64bca (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=B973D714-2FEA7F15 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 1b8b09667e9df557542efb4f14c39cfa@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #130 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '1b8b09667e9df557542efb4f14c39cfa@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK196fc6e8;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as0638f944 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=B948FCC6-859219E1 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 04fe6a451ee8cfa34de303711e6a47dd@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #126 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '04fe6a451ee8cfa34de303711e6a47dd@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4b45e57e;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as4fcef59b (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=BDD8494C-19400333 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 7bc3c69b32b61cbf309192862b790a62@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #128 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '7bc3c69b32b61cbf309192862b790a62@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK458aba33;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as49765fa6 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=9F31ECFB-B95569FC (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 44a157f51cc632a07fcc834d2e48c37a@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #132 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '44a157f51cc632a07fcc834d2e48c37a@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK087ae2c2;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as6e3ccbf8 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=594CB957-2D4FA7A0 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 35d9df0a62eaddc10cd8f6b85ca7ec53@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #138 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '35d9df0a62eaddc10cd8f6b85ca7ec53@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK10cab6bc;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as2002f119 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=27E8840D-90955522 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 266b37c01942fadb6f6119a35230f854@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #134 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '266b37c01942fadb6f6119a35230f854@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK73b15e95;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as7beb00fc (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=A31FD301-F7A53AD0 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 0f3d488f53f529e3087462410ead5b86@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #136 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '0f3d488f53f529e3087462410ead5b86@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK5bb5f849;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as6046627f (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=1EC074DF-3F0D5A22 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 7d00a3a75faca130151daa8d0a13f657@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #140 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '7d00a3a75faca130151daa8d0a13f657@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6723bd76;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as56d90401 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4F43E2AC-BB54A14F (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 097a0aa53d977bf16122c0500125b16a@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #144 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '097a0aa53d977bf16122c0500125b16a@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6c229729;rport (60) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 2: From: "asterisk" ;tag=as343f2592 (56) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=B584C4AB-72B088D4 (53) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 OPTIONS (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 562265857c5e6d886df63907376c98ad@10.0.5.15 (51) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:25 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #142 Oct 1 21:01:25 DEBUG[10546] chan_sip.c: Stopping retransmission on '562265857c5e6d886df63907376c98ad@10.0.5.15' of Request 102: Match Found Oct 1 21:01:25 DEBUG[10566] pbx.c: Launching 'Set' Oct 1 21:01:25 VERBOSE[10566] logger.c: -- Executing Set("Zap/1-1", "TIMEOUT(digit)=5") in new stack Oct 1 21:01:25 VERBOSE[10566] logger.c: -- Digit timeout set to 5 Oct 1 21:01:25 DEBUG[10566] pbx.c: Launching 'Set' Oct 1 21:01:25 VERBOSE[10566] logger.c: -- Executing Set("Zap/1-1", "TIMEOUT(response)=10") in new stack Oct 1 21:01:25 VERBOSE[10566] logger.c: -- Response timeout set to 10 Oct 1 21:01:25 DEBUG[10566] pbx.c: Launching 'SetMusicOnHold' Oct 1 21:01:25 VERBOSE[10566] logger.c: -- Executing SetMusicOnHold("Zap/1-1", "default") in new stack Oct 1 21:01:25 DEBUG[10566] pbx.c: Launching 'BackGround' Oct 1 21:01:25 VERBOSE[10566] logger.c: -- Executing BackGround("Zap/1-1", "local/welcome") in new stack Oct 1 21:01:25 DEBUG[10566] channel.c: Set channel Zap/1-1 to write format gsm Oct 1 21:01:25 DEBUG[10566] channel.c: Scheduling timer at 160 sample intervals Oct 1 21:01:25 VERBOSE[10566] logger.c: -- Playing 'local/welcome' (language 'en') Oct 1 21:01:29 DEBUG[10566] channel.c: Scheduling timer at 0 sample intervals Oct 1 21:01:29 DEBUG[10566] channel.c: Scheduling timer at 0 sample intervals Oct 1 21:01:29 DEBUG[10566] channel.c: Set channel Zap/1-1 to write format alaw Oct 1 21:01:29 DEBUG[10566] pbx.c: Launching 'NoOp' Oct 1 21:01:29 VERBOSE[10566] logger.c: -- Executing NoOp("Zap/1-1", "") in new stack Oct 1 21:01:29 DEBUG[10566] pbx.c: Launching 'Queue' Oct 1 21:01:29 VERBOSE[10566] logger.c: -- Executing Queue("Zap/1-1", "queue") in new stack Oct 1 21:01:29 DEBUG[10566] app_queue.c: NO QUEUE_PRIO variable found. Using default. Oct 1 21:01:29 DEBUG[10566] app_queue.c: queue: queue, options: (null), url: (null), announce: (null), expires: 0, priority: 0 Oct 1 21:01:29 DEBUG[10566] app_queue.c: Queue 'queue' Join, Channel 'Zap/1-1', Position '1' Oct 1 21:01:29 VERBOSE[10566] logger.c: -- Started music on hold, class 'default', on Zap/1-1 Oct 1 21:01:29 DEBUG[10566] channel.c: Scheduling timer at 160 sample intervals Oct 1 21:01:29 DEBUG[10566] app_queue.c: It's our turn (Zap/1-1). Oct 1 21:01:29 DEBUG[10566] app_queue.c: Zap/1-1 is trying to call a queue member. Oct 1 21:01:29 DEBUG[10566] app_queue.c: Simple queue (no URL) Oct 1 21:01:29 DEBUG[10566] app_queue.c: Simple queue (no URL) Oct 1 21:01:29 DEBUG[10566] app_queue.c: Trying 'Agent/6108' with metric 0 Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-open-s-9. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-open-s-8. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-open-s-7. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-open-s-6. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-open-s-5. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-open-s-4. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-open-s-3. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-open-s-2. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-open-s-1. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable STACK-remote-85661098-1. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable CALLEDTON. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable ANI2. Oct 1 21:01:29 DEBUG[10566] channel.c: Not copying variable TRANSFERCAPABILITY. Oct 1 21:01:29 VERBOSE[10566] logger.c: -- outgoing agentcall, to agent '6108', on 'Local/6108@agents-7433,1' Oct 1 21:01:29 DEBUG[10538] devicestate.c: Changing state for Local/6108@agents - state 2 (In use) Oct 1 21:01:29 VERBOSE[10566] logger.c: -- Called Agent/6108 Oct 1 21:01:29 DEBUG[10566] channel.c: Generator got voice, switching to phase locked mode Oct 1 21:01:29 DEBUG[10566] channel.c: Scheduling timer at 0 sample intervals Oct 1 21:01:29 DEBUG[10566] channel.c: Set channel Zap/1-1 to write format slin Oct 1 21:01:29 DEBUG[10566] res_musiconhold.c: Zap/1-1 Opened file 4 '/home/asterisk/mohmp3/Track5' Oct 1 21:01:29 WARNING[10566] interface.c: Junk at the beginning of frame 49443303 Oct 1 21:01:29 DEBUG[10571] pbx.c: Function result is 'SIP/polycom_i1' Oct 1 21:01:29 DEBUG[10571] pbx.c: Launching 'Set' Oct 1 21:01:29 VERBOSE[10571] logger.c: -- Executing Set("Local/6108@agents-7433,2", "chan=SIP/polycom_i1") in new stack Oct 1 21:01:29 DEBUG[10571] pbx.c: Launching 'Set' Oct 1 21:01:29 VERBOSE[10571] logger.c: -- Executing Set("Local/6108@agents-7433,2", "GROUP()=6108") in new stack Oct 1 21:01:29 DEBUG[10571] pbx.c: Function result is '1' Oct 1 21:01:29 DEBUG[10571] pbx.c: Expression result is '0' Oct 1 21:01:29 DEBUG[10571] pbx.c: Launching 'GotoIf' Oct 1 21:01:29 VERBOSE[10571] logger.c: -- Executing GotoIf("Local/6108@agents-7433,2", "0?busy") in new stack Oct 1 21:01:29 DEBUG[10571] pbx.c: Not taking any branch Oct 1 21:01:29 DEBUG[10571] pbx.c: Launching 'Dial' Oct 1 21:01:29 VERBOSE[10571] logger.c: -- Executing Dial("Local/6108@agents-7433,2", "SIP/polycom_i1||20") in new stack Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Setting NAT on RTP to 0 Oct 1 21:01:29 DEBUG[10571] channel.c: Not copying variable STACK-agents-6108-4. Oct 1 21:01:29 DEBUG[10571] channel.c: Not copying variable STACK-agents-6108-3. Oct 1 21:01:29 DEBUG[10571] channel.c: Not copying variable GROUP. Oct 1 21:01:29 DEBUG[10571] channel.c: Not copying variable STACK-agents-6108-2. Oct 1 21:01:29 DEBUG[10571] channel.c: Not copying variable chan. Oct 1 21:01:29 DEBUG[10571] channel.c: Not copying variable STACK-agents-6108-1. Oct 1 21:01:29 DEBUG[10571] channel.c: Not copying variable DB_RESULT. Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Outgoing Call for polycom_i1 Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Updating call counter for outgoing call Oct 1 21:01:29 VERBOSE[10571] logger.c: We're at 10.0.5.15 port 12242 Oct 1 21:01:29 VERBOSE[10571] logger.c: Adding codec 0x8 (alaw) to SDP Oct 1 21:01:29 VERBOSE[10571] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 0: INVITE sip:polycom_i1@10.0.5.133 SIP/2.0 (40) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4937fb5e;rport (60) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 3: To: (31) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 4: Contact: (35) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:29 GMT (35) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 12: Content-Length: 210 (19) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 13: (0) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: v=0 (3) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: o=root 10571 10571 IN IP4 10.0.5.15 (35) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: s=session (9) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: m=audio 12242 RTP/AVP 8 101 (27) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:29 VERBOSE[10571] logger.c: 13 headers, 10 lines Oct 1 21:01:29 VERBOSE[10571] logger.c: Reliably Transmitting (no NAT) to 10.0.5.133:5060: INVITE sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4937fb5e;rport From: "0285661000" ;tag=as64d4e590 To: Contact: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 01 Oct 2006 11:01:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 210 v=0 o=root 10571 10571 IN IP4 10.0.5.15 s=session c=IN IP4 10.0.5.15 t=0 0 m=audio 12242 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 0: INVITE sip:polycom_i1@10.0.5.133 SIP/2.0 (40) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4937fb5e;rport (60) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 3: To: (31) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 4: Contact: (35) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:29 GMT (35) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 12: Content-Length: 210 (19) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Header 13: (0) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: v=0 (3) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: o=root 10571 10571 IN IP4 10.0.5.15 (35) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: s=session (9) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: m=audio 12242 RTP/AVP 8 101 (27) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:29 DEBUG[10571] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #162 Oct 1 21:01:29 VERBOSE[10571] logger.c: -- Called polycom_i1 Oct 1 21:01:29 DEBUG[10571] channel.c: Set channel SIP/polycom_i1-081f9a60 to read format slin Oct 1 21:01:29 DEBUG[10571] channel.c: Set channel SIP/polycom_i1-081f9a60 to write format slin Oct 1 21:01:29 DEBUG[10572] app_queue.c: Device 'Local/6108@agents' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:29 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4937fb5e;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 CSeq: 102 INVITE Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Content-Length: 0 Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4937fb5e;rport (60) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 8: Content-Length: 0 (17) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 9: (0) Oct 1 21:01:29 VERBOSE[10546] logger.c: --- (9 headers 0 lines) --- Oct 1 21:01:29 DEBUG[10546] chan_sip.c: *** SIP TIMER: Cancelling retransmission #162 - INVITE (got response) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5cc79f12017802bd142be41c23692cab@10.0.5.15' Request 102: Found Oct 1 21:01:29 DEBUG[10546] chan_sip.c: SIP response 100 to standard invite Oct 1 21:01:29 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4937fb5e;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 CSeq: 102 INVITE Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Allow-Events: talk,hold,conference Content-Length: 0 Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4937fb5e;rport (60) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 8: Allow-Events: talk,hold,conference (34) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:29 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:29 VERBOSE[10546] logger.c: --- (10 headers 0 lines) --- Oct 1 21:01:29 DEBUG[10546] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5cc79f12017802bd142be41c23692cab@10.0.5.15' Request 102: Found Oct 1 21:01:29 DEBUG[10546] chan_sip.c: SIP response 180 to standard invite Oct 1 21:01:29 VERBOSE[10571] logger.c: -- SIP/polycom_i1-081f9a60 is ringing Oct 1 21:01:29 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:29 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_i1 - state 6 (Ringing) Oct 1 21:01:29 VERBOSE[10566] logger.c: -- Agent/6108 is ringing Oct 1 21:01:29 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:29 DEBUG[10573] app_queue.c: Device 'SIP/polycom_i1' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Oct 1 21:01:31 DEBUG[10562] rtp.c: Got RTCP report of 100 bytes Oct 1 21:01:33 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4937fb5e;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 CSeq: 102 INVITE Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1159700490 1159700490 IN IP4 10.0.5.133 s=Polycom IP Phone c=IN IP4 10.0.5.133 t=0 0 m=audio 2236 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK4937fb5e;rport (60) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 183 (19) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Line: o=- 1159700490 1159700490 IN IP4 10.0.5.133 (43) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Line: s=Polycom IP Phone (18) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Line: c=IN IP4 10.0.5.133 (19) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Line: m=audio 2236 RTP/AVP 8 101 (26) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:33 VERBOSE[10546] logger.c: --- (11 headers 8 lines) --- Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Acked pending invite 102 Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Stopping retransmission on '5cc79f12017802bd142be41c23692cab@10.0.5.15' of Request 102: Match Found Oct 1 21:01:33 DEBUG[10546] chan_sip.c: SIP response 200 to standard invite Oct 1 21:01:33 VERBOSE[10546] logger.c: Found RTP audio format 8 Oct 1 21:01:33 VERBOSE[10546] logger.c: Found RTP audio format 101 Oct 1 21:01:33 VERBOSE[10546] logger.c: Peer audio RTP is at port 10.0.5.133:2236 Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Peer audio RTP is at port 10.0.5.133:2236 Oct 1 21:01:33 VERBOSE[10546] logger.c: Found description format PCMA Oct 1 21:01:33 VERBOSE[10546] logger.c: Found description format telephone-event Oct 1 21:01:33 VERBOSE[10546] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Oct 1 21:01:33 VERBOSE[10546] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: build_route: Contact hop: Oct 1 21:01:33 VERBOSE[10546] logger.c: list_route: hop: Oct 1 21:01:33 VERBOSE[10546] logger.c: set_destination: Parsing for address/port to send to Oct 1 21:01:33 VERBOSE[10546] logger.c: set_destination: set destination to 10.0.5.133, port 5060 Oct 1 21:01:33 VERBOSE[10546] logger.c: Transmitting (no NAT) to 10.0.5.133:5060: ACK sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK36e5438e;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 Contact: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 0: ACK sip:polycom_i1@10.0.5.133 SIP/2.0 (37) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK36e5438e;rport (60) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 4: Contact: (35) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 ACK (13) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:33 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:33 VERBOSE[10571] logger.c: -- SIP/polycom_i1-081f9a60 answered Local/6108@agents-7433,2 Oct 1 21:01:33 DEBUG[10566] app_queue.c: Dunno what to do with control type -1 Oct 1 21:01:33 VERBOSE[10566] logger.c: -- Agent/6108 answered Zap/1-1 Oct 1 21:01:33 DEBUG[10566] app_queue.c: Next is 'Agent/6109' with metric 1 Oct 1 21:01:33 DEBUG[10566] chan_zap.c: Set option TONE VERIFY, mode: MUTECONF(1) on Zap/1-1 Oct 1 21:01:33 VERBOSE[10566] logger.c: -- Stopped music on hold on Zap/1-1 Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel Zap/1-1 to write format alaw Oct 1 21:01:33 DEBUG[10566] channel.c: Scheduling timer at 0 sample intervals Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel Zap/1-1 to read format slin Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel Zap/1-1 to write format slin Oct 1 21:01:33 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:33 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_i1 - state 2 (In use) Oct 1 21:01:33 DEBUG[10571] rtp.c: Ooh, format changed from unknown to alaw Oct 1 21:01:33 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:33 DEBUG[10538] devicestate.c: Changing state for Local/6108@agents - state 2 (In use) Oct 1 21:01:33 DEBUG[10538] devicestate.c: Changing state for Agent/6108 - state 3 (Busy) Oct 1 21:01:33 DEBUG[10538] devicestate.c: Changing state for Local/6108@agents - state 2 (In use) Oct 1 21:01:33 DEBUG[10574] app_queue.c: Device 'SIP/polycom_i1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:33 DEBUG[10575] app_queue.c: Device 'Local/6108@agents' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:33 DEBUG[10576] app_queue.c: Device 'Agent/6108' changed to state '3' (Busy) Oct 1 21:01:33 DEBUG[10577] app_queue.c: Device 'Local/6108@agents' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:33 DEBUG[10571] channel.c: Planning to masquerade channel SIP/polycom_i1-081f9a60 into the structure of Local/6108@agents-7433,1 Oct 1 21:01:33 DEBUG[10571] channel.c: Done planning to masquerade channel SIP/polycom_i1-081f9a60 into the structure of Local/6108@agents-7433,1 Oct 1 21:01:33 DEBUG[10571] chan_local.c: Not posting to queue since already masked on 'Local/6108@agents-7433,2' Oct 1 21:01:33 DEBUG[10566] channel.c: Actually Masquerading SIP/polycom_i1-081f9a60(6) into the structure of Local/6108@agents-7433,1(6) Oct 1 21:01:33 DEBUG[10566] channel.c: Got clone lock for masquerade on 'SIP/polycom_i1-081f9a60' at 0x821b1ec Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel SIP/polycom_i1-081f9a60 to write format slin Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel SIP/polycom_i1-081f9a60 to read format slin Oct 1 21:01:33 DEBUG[10566] channel.c: Putting channel SIP/polycom_i1-081f9a60 in 64/64 formats Oct 1 21:01:33 DEBUG[10566] channel.c: Released clone lock on 'Local/6108@agents-7433,1' Oct 1 21:01:33 DEBUG[10566] channel.c: Done Masquerading SIP/polycom_i1-081f9a60 (6) Oct 1 21:01:33 DEBUG[10566] chan_agent.c: Bridge on 'SIP/polycom_i1-081f9a60' being set to 'Agent/6108' (3) Oct 1 21:01:33 DEBUG[10566] chan_agent.c: Native formats changing from 64 to 8 Oct 1 21:01:33 DEBUG[10566] chan_agent.c: Resetting read to 64 and write to 64 Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel Agent/6108 to read format slin Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel Agent/6108 to write format slin Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel SIP/polycom_i1-081f9a60 to read format alaw Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel SIP/polycom_i1-081f9a60 to write format alaw Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel Zap/1-1 to read format alaw Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel Agent/6108 to write format alaw Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel Agent/6108 to read format alaw Oct 1 21:01:33 DEBUG[10566] channel.c: Set channel Zap/1-1 to write format alaw Oct 1 21:01:33 DEBUG[10571] channel.c: Didn't get a frame from channel: Local/6108@agents-7433,1 Oct 1 21:01:33 DEBUG[10571] channel.c: Bridge stops bridging channels Local/6108@agents-7433,2 and Local/6108@agents-7433,1 Oct 1 21:01:33 DEBUG[10571] channel.c: Hanging up zombie 'Local/6108@agents-7433,1' Oct 1 21:01:33 DEBUG[10571] app_dial.c: Exiting with DIALSTATUS=ANSWER. Oct 1 21:01:33 DEBUG[10571] pbx.c: Spawn extension (agents,6108,4) exited non-zero on 'Local/6108@agents-7433,2' Oct 1 21:01:33 VERBOSE[10571] logger.c: == Spawn extension (agents, 6108, 4) exited non-zero on 'Local/6108@agents-7433,2' Oct 1 21:01:33 DEBUG[10571] channel.c: Hanging up channel 'Local/6108@agents-7433,2' Oct 1 21:01:33 DEBUG[10538] devicestate.c: Changing state for Local/6108@agents - state 0 (Unknown) Oct 1 21:01:33 DEBUG[10538] devicestate.c: Changing state for Local/6108@agents - state 0 (Unknown) Oct 1 21:01:33 DEBUG[10578] app_queue.c: Device 'Local/6108@agents' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Oct 1 21:01:33 DEBUG[10579] app_queue.c: Device 'Local/6108@agents' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Oct 1 21:01:39 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: INVITE sip:0285661000@10.0.5.15 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK13ca06749E2CD0D5 From: ;tag=4BD59B6F-4FA1ECB6 To: "0285661000" ;tag=as64d4e590 CSeq: 1 INVITE Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 180 v=0 o=- 1159700490 1159700491 IN IP4 10.0.5.133 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2236 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 0: INVITE sip:0285661000@10.0.5.15 SIP/2.0 (39) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK13ca06749E2CD0D5 (58) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=4BD59B6F-4FA1ECB6 (55) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 3: To: "0285661000" ;tag=as64d4e590 (58) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 4: CSeq: 1 INVITE (14) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 9: Supported: 100rel,replaces (26) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 11: Max-Forwards: 70 (16) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 12: Content-Type: application/sdp (29) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 13: Content-Length: 180 (19) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 14: (0) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: o=- 1159700490 1159700491 IN IP4 10.0.5.133 (43) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: s=Polycom IP Phone (18) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: c=IN IP4 0.0.0.0 (16) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: m=audio 2236 RTP/AVP 8 101 (26) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:39 VERBOSE[10546] logger.c: --- (14 headers 8 lines) --- Oct 1 21:01:39 DEBUG[10546] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Begin: parsing SIP "Supported: 100rel,replaces" Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Found SIP option: -100rel- Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Matched SIP option: 100rel Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Found SIP option: -replaces- Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Matched SIP option: replaces Oct 1 21:01:39 DEBUG[10546] chan_sip.c: * SIP extension value: 3 for call 5cc79f12017802bd142be41c23692cab@10.0.5.15 Oct 1 21:01:39 VERBOSE[10546] logger.c: Using INVITE request as basis request - 5cc79f12017802bd142be41c23692cab@10.0.5.15 Oct 1 21:01:39 VERBOSE[10546] logger.c: Sending to 10.0.5.133 : 5060 (non-NAT) Oct 1 21:01:39 VERBOSE[10546] logger.c: Found RTP audio format 8 Oct 1 21:01:39 VERBOSE[10546] logger.c: Found RTP audio format 101 Oct 1 21:01:39 VERBOSE[10546] logger.c: Peer audio RTP is at port 0.0.0.0:2236 Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Peer audio RTP is at port 0.0.0.0:2236 Oct 1 21:01:39 VERBOSE[10546] logger.c: Found description format PCMA Oct 1 21:01:39 VERBOSE[10546] logger.c: Found description format telephone-event Oct 1 21:01:39 VERBOSE[10546] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Oct 1 21:01:39 VERBOSE[10546] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Oct 1 21:01:39 DEBUG[10546] chan_agent.c: Asked for bridged channel on 'SIP/polycom_i1-081f9a60'/'Agent/6108', returning 'Zap/1-1' Oct 1 21:01:39 VERBOSE[10546] logger.c: -- Started music on hold, class 'default', on Zap/1-1 Oct 1 21:01:39 DEBUG[10546] channel.c: Scheduling timer at 160 sample intervals Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Got a SIP re-invite for call 5cc79f12017802bd142be41c23692cab@10.0.5.15 Oct 1 21:01:39 VERBOSE[10546] logger.c: We're at 10.0.5.15 port 12242 Oct 1 21:01:39 VERBOSE[10546] logger.c: Adding codec 0x8 (alaw) to SDP Oct 1 21:01:39 VERBOSE[10546] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Oct 1 21:01:39 VERBOSE[10546] logger.c: Reliably Transmitting (no NAT) to 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK13ca06749E2CD0D5;received=10.0.5.133 From: ;tag=4BD59B6F-4FA1ECB6 To: "0285661000" ;tag=as64d4e590 Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 210 v=0 o=root 10571 10572 IN IP4 10.0.5.15 s=session c=IN IP4 10.0.5.15 t=0 0 m=audio 12242 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK13ca06749E2CD0D5;received=10.0.5.133 (78) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=4BD59B6F-4FA1ECB6 (55) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 3: To: "0285661000" ;tag=as64d4e590 (58) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 4: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 5: CSeq: 1 INVITE (14) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 8: Contact: (35) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 210 (19) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: o=root 10571 10572 IN IP4 10.0.5.15 (35) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: s=session (9) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: m=audio 12242 RTP/AVP 8 101 (27) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #164 Oct 1 21:01:39 DEBUG[10566] channel.c: Generator got voice, switching to phase locked mode Oct 1 21:01:39 DEBUG[10566] channel.c: Scheduling timer at 0 sample intervals Oct 1 21:01:39 DEBUG[10566] channel.c: Set channel Zap/1-1 to write format slin Oct 1 21:01:39 DEBUG[10566] res_musiconhold.c: Zap/1-1 Opened file 4 '/home/asterisk/mohmp3/Track5' Oct 1 21:01:39 WARNING[10566] interface.c: Junk at the beginning of frame 49443303 Oct 1 21:01:39 DEBUG[10546] chan_sip.c: SIP TIMER: Rescheduling retransmission #164 (1) SIP/2.0 - 1 Oct 1 21:01:39 DEBUG[10546] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 84 ms (t1 42 ms (Retrans id #164)) Oct 1 21:01:39 VERBOSE[10546] logger.c: Retransmitting #1 (no NAT) to 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK13ca06749E2CD0D5;received=10.0.5.133 From: ;tag=4BD59B6F-4FA1ECB6 To: "0285661000" ;tag=as64d4e590 Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 210 v=0 o=root 10571 10572 IN IP4 10.0.5.15 s=session c=IN IP4 10.0.5.15 t=0 0 m=audio 12242 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Oct 1 21:01:39 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: ACK sip:0285661000@10.0.5.15 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK54feac622F4342AB From: ;tag=4BD59B6F-4FA1ECB6 To: "0285661000" ;tag=as64d4e590 CSeq: 1 ACK Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Max-Forwards: 70 Content-Length: 0 Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 0: ACK sip:0285661000@10.0.5.15 SIP/2.0 (36) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK54feac622F4342AB (58) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=4BD59B6F-4FA1ECB6 (55) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 3: To: "0285661000" ;tag=as64d4e590 (58) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 4: CSeq: 1 ACK (11) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 9: Max-Forwards: 70 (16) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 0 (17) Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:39 VERBOSE[10546] logger.c: --- (11 headers 0 lines) --- Oct 1 21:01:39 DEBUG[10546] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Oct 1 21:01:39 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #164 Oct 1 21:01:39 DEBUG[10546] chan_sip.c: Stopping retransmission on '5cc79f12017802bd142be41c23692cab@10.0.5.15' of Response 1: Match Found Oct 1 21:01:41 DEBUG[10562] rtp.c: Got RTCP report of 100 bytes Oct 1 21:01:41 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: INVITE sip:6109@10.0.5.15:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK50b9aa00127B7671 From: "Line1" ;tag=A3D937FA-92CA85CD To: CSeq: 1 INVITE Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 233 v=0 o=- 1159700498 1159700498 IN IP4 10.0.5.133 s=Polycom IP Phone c=IN IP4 10.0.5.133 t=0 0 m=audio 2226 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 0: INVITE sip:6109@10.0.5.15:5060;user=phone SIP/2.0 (49) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK50b9aa00127B7671 (58) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 2: From: "Line1" ;tag=A3D937FA-92CA85CD (62) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 3: To: (35) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 4: CSeq: 1 INVITE (14) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 5: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 9: Supported: 100rel,replaces (26) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 10: Allow-Events: talk,hold,conference (34) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 11: Max-Forwards: 70 (16) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 12: Content-Type: application/sdp (29) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 13: Content-Length: 233 (19) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 14: (0) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: o=- 1159700498 1159700498 IN IP4 10.0.5.133 (43) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: s=Polycom IP Phone (18) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: c=IN IP4 10.0.5.133 (19) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 0 18 101 (31) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:41 VERBOSE[10546] logger.c: --- (14 headers 10 lines) --- Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for f61336ce-c583d64c-8566eda7@10.0.5.133 - INVITE (With RTP) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Begin: parsing SIP "Supported: 100rel,replaces" Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Found SIP option: -100rel- Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Matched SIP option: 100rel Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Found SIP option: -replaces- Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Matched SIP option: replaces Oct 1 21:01:41 DEBUG[10546] chan_sip.c: * SIP extension value: 3 for call f61336ce-c583d64c-8566eda7@10.0.5.133 Oct 1 21:01:41 VERBOSE[10546] logger.c: Using INVITE request as basis request - f61336ce-c583d64c-8566eda7@10.0.5.133 Oct 1 21:01:41 VERBOSE[10546] logger.c: Sending to 10.0.5.133 : 5060 (non-NAT) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Setting NAT on RTP to 0 Oct 1 21:01:41 VERBOSE[10546] logger.c: Found user 'polycom_i1' Oct 1 21:01:41 VERBOSE[10546] logger.c: Found RTP audio format 8 Oct 1 21:01:41 VERBOSE[10546] logger.c: Found RTP audio format 0 Oct 1 21:01:41 VERBOSE[10546] logger.c: Found RTP audio format 18 Oct 1 21:01:41 VERBOSE[10546] logger.c: Found RTP audio format 101 Oct 1 21:01:41 VERBOSE[10546] logger.c: Peer audio RTP is at port 10.0.5.133:2226 Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Peer audio RTP is at port 10.0.5.133:2226 Oct 1 21:01:41 VERBOSE[10546] logger.c: Found description format PCMA Oct 1 21:01:41 VERBOSE[10546] logger.c: Found description format PCMU Oct 1 21:01:41 VERBOSE[10546] logger.c: Found description format G729 Oct 1 21:01:41 VERBOSE[10546] logger.c: Found description format telephone-event Oct 1 21:01:41 VERBOSE[10546] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Oct 1 21:01:41 VERBOSE[10546] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Checking SIP call limits for device polycom_i1 Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Updating call counter for incoming call Oct 1 21:01:41 VERBOSE[10546] logger.c: Looking for 6109 in inside (domain 10.0.5.15) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: build_route: Contact hop: Oct 1 21:01:41 VERBOSE[10546] logger.c: list_route: hop: Oct 1 21:01:41 VERBOSE[10546] logger.c: Transmitting (no NAT) to 10.0.5.133:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK50b9aa00127B7671;received=10.0.5.133 From: "Line1" ;tag=A3D937FA-92CA85CD To: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK50b9aa00127B7671;received=10.0.5.133 (78) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 2: From: "Line1" ;tag=A3D937FA-92CA85CD (62) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 3: To: (35) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 4: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 5: CSeq: 1 INVITE (14) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 8: Contact: (29) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:41 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:41 DEBUG[10538] channel.c: Avoiding initial deadlock for 'SIP/polycom_i1-081d86d8' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Macro' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Macro("SIP/polycom_i1-081d86d8", "stdexten|6109") in new stack Oct 1 21:01:41 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_i1 - state 2 (In use) Oct 1 21:01:41 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:41 DEBUG[10580] pbx.c: Function result is 'SIP/polycom_i1' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Set' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Set("SIP/polycom_i1-081d86d8", "member=SIP/polycom_i1") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Set' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Set("SIP/polycom_i1-081d86d8", "GROUP()=SIP/polycom_i1") in new stack Oct 1 21:01:41 DEBUG[10580] db.c: Unable to find key '6109' in family 'DND' Oct 1 21:01:41 DEBUG[10580] func_db.c: DB: DND/6109 not found in database. Oct 1 21:01:41 DEBUG[10580] pbx.c: Function result is '' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Set' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Set("SIP/polycom_i1-081d86d8", "dnd=") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Expression result is '0' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'GotoIf' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing GotoIf("SIP/polycom_i1-081d86d8", "0?dnd") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Not taking any branch Oct 1 21:01:41 DEBUG[10580] pbx.c: Function result is 'SIP/polycom_j1' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Set' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Set("SIP/polycom_i1-081d86d8", "chan=SIP/polycom_j1") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Expression result is '0' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'GotoIf' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing GotoIf("SIP/polycom_i1-081d86d8", "0?vmbusy") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Not taking any branch Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Set' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Set("SIP/polycom_i1-081d86d8", "ATTEMPT=1") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Expression result is '0' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'GotoIf' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing GotoIf("SIP/polycom_i1-081d86d8", "0?loop") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Not taking any branch Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Answer' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Answer("SIP/polycom_i1-081d86d8", "") in new stack Oct 1 21:01:41 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:41 DEBUG[10538] channel.c: Avoiding initial deadlock for 'SIP/polycom_i1-081d86d8' Oct 1 21:01:41 DEBUG[10538] channel.c: Avoiding initial deadlock for 'SIP/polycom_i1-081d86d8' Oct 1 21:01:41 DEBUG[10580] chan_sip.c: sip_answer(SIP/polycom_i1-081d86d8) Oct 1 21:01:41 VERBOSE[10580] logger.c: We're at 10.0.5.15 port 13742 Oct 1 21:01:41 VERBOSE[10580] logger.c: Adding codec 0x8 (alaw) to SDP Oct 1 21:01:41 VERBOSE[10580] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Oct 1 21:01:41 VERBOSE[10580] logger.c: Reliably Transmitting (no NAT) to 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK50b9aa00127B7671;received=10.0.5.133 From: "Line1" ;tag=A3D937FA-92CA85CD To: ;tag=as49e4c4e8 Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 210 v=0 o=root 10580 10580 IN IP4 10.0.5.15 s=session c=IN IP4 10.0.5.15 t=0 0 m=audio 13742 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK50b9aa00127B7671;received=10.0.5.133 (78) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 2: From: "Line1" ;tag=A3D937FA-92CA85CD (62) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 3: To: ;tag=as49e4c4e8 (50) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 4: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 5: CSeq: 1 INVITE (14) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 8: Contact: (29) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 10: Content-Length: 210 (19) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 11: (0) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: v=0 (3) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: o=root 10580 10580 IN IP4 10.0.5.15 (35) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: s=session (9) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: m=audio 13742 RTP/AVP 8 101 (27) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #165 Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Goto' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Goto("SIP/polycom_i1-081d86d8", "loop") in new stack Oct 1 21:01:41 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_i1 - state 2 (In use) Oct 1 21:01:41 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Goto (macro-stdexten,s,11) Oct 1 21:01:41 DEBUG[10580] pbx.c: Function result is 'SIP/polycom_j1' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Set' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Set("SIP/polycom_i1-081d86d8", "mychan=SIP/polycom_j1") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Expression result is '0' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'GotoIf' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing GotoIf("SIP/polycom_i1-081d86d8", "0?vmbusy") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Not taking any branch Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'ChanIsAvail' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing ChanIsAvail("SIP/polycom_i1-081d86d8", "SIP/polycom_j1") in new stack Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Setting NAT on RTP to 0 Oct 1 21:01:41 DEBUG[10580] channel.c: Hanging up channel 'SIP/polycom_j1-081db2f8' Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Hangup call SIP/polycom_j1-081db2f8, SIP callid 6855a16f56958b6e778b922d49132b72@10.0.5.15) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: update_call_counter(polycom_j1) - decrement call limit counter Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Updating call counter for incoming call Oct 1 21:01:41 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_j1 Oct 1 21:01:41 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_j1 - state 1 (Not in use) Oct 1 21:01:41 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_j1 Oct 1 21:01:41 DEBUG[10580] pbx.c: Expression result is '0' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'GotoIf' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing GotoIf("SIP/polycom_i1-081d86d8", "0?next") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Not taking any branch Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Set' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Set("SIP/polycom_i1-081d86d8", "GROUP()=SIP/polycom_j1") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Function result is '1' Oct 1 21:01:41 DEBUG[10580] pbx.c: Expression result is '0' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'GotoIf' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing GotoIf("SIP/polycom_i1-081d86d8", "0?next") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Not taking any branch Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'MailboxExists' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing MailboxExists("SIP/polycom_i1-081d86d8", "6109") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Expression result is '0' Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'GotoIf' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing GotoIf("SIP/polycom_i1-081d86d8", "0?vmcall:novm") in new stack Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Goto (macro-stdexten,s,19) Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Set' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Set("SIP/polycom_i1-081d86d8", "ring=r") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Set' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Set("SIP/polycom_i1-081d86d8", "time=25") in new stack Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Goto' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Goto("SIP/polycom_i1-081d86d8", "call") in new stack Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Goto (macro-stdexten,s,28) Oct 1 21:01:41 DEBUG[10580] pbx.c: Launching 'Dial' Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Executing Dial("SIP/polycom_i1-081d86d8", "SIP/polycom_j1|25|r") in new stack Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Setting NAT on RTP to 0 Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-28. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable MACRO_DEPTH. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-21. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable time. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-20. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable ring. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-19. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-18. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable VMBOXEXISTSSTATUS. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-17. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-16. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable GROUP. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-15. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-14. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable AVAILSTATUS. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable AVAILORIGCHAN. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable AVAILCHAN. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-13. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-12. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable mychan. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-11. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-10. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-9. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-8. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable ATTEMPT. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-7. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-6. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable chan. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-5. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable DB_RESULT. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-4. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable dnd. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-3. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-2. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable member. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-macro-stdexten-s-1. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable ARG1. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable MACRO_PRIORITY. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable MACRO_CONTEXT. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable MACRO_EXTEN. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable STACK-inside-6109-1. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable SIPCALLID. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable SIPUSERAGENT. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable SIPDOMAIN. Oct 1 21:01:41 DEBUG[10580] channel.c: Not copying variable SIPURI. Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Outgoing Call for polycom_j1 Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Updating call counter for outgoing call Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 0: INVITE sip:polycom_j1@10.0.5.130 SIP/2.0 (40) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK2795383e;rport (60) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 3: To: (31) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:41 GMT (35) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 12: Content-Length: 210 (19) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 13: (0) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: v=0 (3) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: o=root 10580 10580 IN IP4 10.0.5.15 (35) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: s=session (9) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: m=audio 17486 RTP/AVP 8 101 (27) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 0: INVITE sip:polycom_j1@10.0.5.130 SIP/2.0 (40) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK2795383e;rport (60) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 3: To: (31) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 9: Date: Sun, 01 Oct 2006 11:01:41 GMT (35) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 12: Content-Length: 210 (19) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Header 13: (0) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: v=0 (3) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: o=root 10580 10580 IN IP4 10.0.5.15 (35) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: s=session (9) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: m=audio 17486 RTP/AVP 8 101 (27) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:41 DEBUG[10580] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #167 Oct 1 21:01:41 VERBOSE[10580] logger.c: -- Called polycom_j1 Oct 1 21:01:41 DEBUG[10580] channel.c: Driver for channel 'SIP/polycom_i1-081d86d8' does not support indication 3, emulating it Oct 1 21:01:41 DEBUG[10580] channel.c: Set channel SIP/polycom_i1-081d86d8 to write format slin Oct 1 21:01:41 DEBUG[10580] channel.c: Scheduling timer at 160 sample intervals Oct 1 21:01:41 DEBUG[10581] app_queue.c: Device 'SIP/polycom_i1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:41 DEBUG[10582] app_queue.c: Device 'SIP/polycom_i1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:41 DEBUG[10583] app_queue.c: Device 'SIP/polycom_j1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Oct 1 21:01:41 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: ACK sip:6109@10.0.5.15 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK72961758B6419AE9 From: "Line1" ;tag=A3D937FA-92CA85CD To: ;tag=as49e4c4e8 CSeq: 1 ACK Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Max-Forwards: 70 Content-Length: 0 Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 0: ACK sip:6109@10.0.5.15 SIP/2.0 (30) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bK72961758B6419AE9 (58) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 2: From: "Line1" ;tag=A3D937FA-92CA85CD (62) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=as49e4c4e8 (50) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 4: CSeq: 1 ACK (11) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 5: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 9: Max-Forwards: 70 (16) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 0 (17) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:41 VERBOSE[10546] logger.c: --- (11 headers 0 lines) --- Oct 1 21:01:41 DEBUG[10546] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Oct 1 21:01:41 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #165 Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Stopping retransmission on 'f61336ce-c583d64c-8566eda7@10.0.5.133' of Response 1: Match Found Oct 1 21:01:41 DEBUG[10580] rtp.c: Ooh, format changed from unknown to alaw Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 100 Trying (18) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK2795383e;rport (60) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 8: Content-Length: 0 (17) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 9: (0) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: *** SIP TIMER: Cancelling retransmission #167 - INVITE (got response) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5fb12289210feb13773250987390be20@10.0.5.15' Request 102: Found Oct 1 21:01:41 DEBUG[10546] chan_sip.c: SIP response 100 to standard invite Oct 1 21:01:41 DEBUG[10580] channel.c: Generator got voice, switching to phase locked mode Oct 1 21:01:41 DEBUG[10580] channel.c: Scheduling timer at 0 sample intervals Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 180 Ringing (19) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK2795383e;rport (60) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 8: Allow-Events: talk,hold,conference (34) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:41 DEBUG[10546] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5fb12289210feb13773250987390be20@10.0.5.15' Request 102: Found Oct 1 21:01:41 DEBUG[10546] chan_sip.c: SIP response 180 to standard invite Oct 1 21:01:41 VERBOSE[10580] logger.c: -- SIP/polycom_j1-081e0838 is ringing Oct 1 21:01:41 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_j1 Oct 1 21:01:41 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_j1 - state 6 (Ringing) Oct 1 21:01:41 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_j1 Oct 1 21:01:41 DEBUG[10584] app_queue.c: Device 'SIP/polycom_j1' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK2795383e;rport (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 183 (19) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: o=- 1159700501 1159700501 IN IP4 10.0.5.130 (43) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: s=Polycom IP Phone (18) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: c=IN IP4 10.0.5.130 (19) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 101 (26) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Acked pending invite 102 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Stopping retransmission on '5fb12289210feb13773250987390be20@10.0.5.15' of Request 102: Match Found Oct 1 21:01:44 DEBUG[10546] chan_sip.c: SIP response 200 to standard invite Oct 1 21:01:44 DEBUG[10546] chan_sip.c: build_route: Contact hop: Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 0: ACK sip:polycom_j1@10.0.5.130 SIP/2.0 (37) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK3f66b7eb;rport (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 ACK (13) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:44 VERBOSE[10580] logger.c: -- SIP/polycom_j1-081e0838 answered SIP/polycom_i1-081d86d8 Oct 1 21:01:44 DEBUG[10580] channel.c: Set channel SIP/polycom_i1-081d86d8 to write format alaw Oct 1 21:01:44 DEBUG[10580] channel.c: Scheduling timer at 0 sample intervals Oct 1 21:01:44 VERBOSE[10580] logger.c: -- Attempting native bridge of SIP/polycom_i1-081d86d8 and SIP/polycom_j1-081e0838 Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Sending reinvite on SIP 'f61336ce-c583d64c-8566eda7@10.0.5.133' - It's audio soon redirected to IP 10.0.5.130 Oct 1 21:01:44 VERBOSE[10580] logger.c: set_destination: Parsing for address/port to send to Oct 1 21:01:44 VERBOSE[10580] logger.c: set_destination: set destination to 10.0.5.133, port 5060 Oct 1 21:01:44 VERBOSE[10580] logger.c: We're at 10.0.5.15 port 13742 Oct 1 21:01:44 VERBOSE[10580] logger.c: Adding codec 0x8 (alaw) to SDP Oct 1 21:01:44 VERBOSE[10580] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 0: INVITE sip:polycom_i1@10.0.5.133 SIP/2.0 (40) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK211c26d1;rport (60) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 2: From: ;tag=as49e4c4e8 (52) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 3: To: "Line1" ;tag=A3D937FA-92CA85CD (60) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 5: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 12: Content-Length: 211 (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 13: (0) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: v=0 (3) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: o=root 10580 10581 IN IP4 10.0.5.130 (36) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: s=session (9) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: c=IN IP4 10.0.5.130 (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 101 (26) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:44 VERBOSE[10580] logger.c: 13 headers, 10 lines Oct 1 21:01:44 VERBOSE[10580] logger.c: Reliably Transmitting (no NAT) to 10.0.5.133:5060: INVITE sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK211c26d1;rport From: ;tag=as49e4c4e8 To: "Line1" ;tag=A3D937FA-92CA85CD Contact: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 211 v=0 o=root 10580 10581 IN IP4 10.0.5.130 s=session c=IN IP4 10.0.5.130 t=0 0 m=audio 2226 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 0: INVITE sip:polycom_i1@10.0.5.133 SIP/2.0 (40) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK211c26d1;rport (60) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 2: From: ;tag=as49e4c4e8 (52) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 3: To: "Line1" ;tag=A3D937FA-92CA85CD (60) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 5: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 6: CSeq: 102 INVITE (16) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 12: Content-Length: 211 (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 13: (0) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: v=0 (3) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: o=root 10580 10581 IN IP4 10.0.5.130 (36) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: s=session (9) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: c=IN IP4 10.0.5.130 (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 101 (26) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #169 Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Sending reinvite on SIP '5fb12289210feb13773250987390be20@10.0.5.15' - It's audio soon redirected to IP 10.0.5.133 Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 0: INVITE sip:polycom_j1@10.0.5.130 SIP/2.0 (40) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK7a91ba89;rport (60) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 6: CSeq: 103 INVITE (16) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 12: Content-Length: 282 (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 13: (0) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: v=0 (3) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: o=root 10580 10581 IN IP4 10.0.5.133 (36) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: s=session (9) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: c=IN IP4 10.0.5.133 (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 0 18 101 (31) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=fmtp:18 annexb=no (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 0: INVITE sip:polycom_j1@10.0.5.130 SIP/2.0 (40) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK7a91ba89;rport (60) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 6: CSeq: 103 INVITE (16) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 12: Content-Length: 282 (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Header 13: (0) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: v=0 (3) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: o=root 10580 10581 IN IP4 10.0.5.133 (36) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: s=session (9) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: c=IN IP4 10.0.5.133 (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 0 18 101 (31) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=fmtp:18 annexb=no (19) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:44 DEBUG[10580] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #170 Oct 1 21:01:44 DEBUG[10580] rtp.c: Ooh, format changed from unknown to alaw Oct 1 21:01:44 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_j1 Oct 1 21:01:44 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_j1 - state 2 (In use) Oct 1 21:01:44 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_j1 Oct 1 21:01:44 DEBUG[10585] app_queue.c: Device 'SIP/polycom_j1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:44 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK211c26d1;rport From: ;tag=as49e4c4e8 To: "Line1" ;tag=A3D937FA-92CA85CD CSeq: 102 INVITE Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1159700498 1159700499 IN IP4 10.0.5.133 s=Polycom IP Phone c=IN IP4 10.0.5.133 t=0 0 m=audio 2226 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK211c26d1;rport (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=as49e4c4e8 (52) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 3: To: "Line1" ;tag=A3D937FA-92CA85CD (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 INVITE (16) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 5: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 183 (19) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: o=- 1159700498 1159700499 IN IP4 10.0.5.133 (43) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: s=Polycom IP Phone (18) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: c=IN IP4 10.0.5.133 (19) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 101 (26) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:44 VERBOSE[10546] logger.c: --- (11 headers 8 lines) --- Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Acked pending invite 102 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #169 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Stopping retransmission on 'f61336ce-c583d64c-8566eda7@10.0.5.133' of Request 102: Match Found Oct 1 21:01:44 DEBUG[10546] chan_sip.c: SIP response 200 to RE-invite on outgoing call f61336ce-c583d64c-8566eda7@10.0.5.133 Oct 1 21:01:44 VERBOSE[10546] logger.c: Found RTP audio format 8 Oct 1 21:01:44 VERBOSE[10546] logger.c: Found RTP audio format 101 Oct 1 21:01:44 VERBOSE[10546] logger.c: Peer audio RTP is at port 10.0.5.133:2226 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Peer audio RTP is at port 10.0.5.133:2226 Oct 1 21:01:44 VERBOSE[10546] logger.c: Found description format PCMA Oct 1 21:01:44 VERBOSE[10546] logger.c: Found description format telephone-event Oct 1 21:01:44 VERBOSE[10546] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Oct 1 21:01:44 VERBOSE[10546] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: build_route: Contact hop: Oct 1 21:01:44 VERBOSE[10546] logger.c: list_route: hop: Oct 1 21:01:44 VERBOSE[10546] logger.c: set_destination: Parsing for address/port to send to Oct 1 21:01:44 VERBOSE[10546] logger.c: set_destination: set destination to 10.0.5.133, port 5060 Oct 1 21:01:44 VERBOSE[10546] logger.c: Transmitting (no NAT) to 10.0.5.133:5060: ACK sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK3dcc37b2;rport From: ;tag=as49e4c4e8 To: "Line1" ;tag=A3D937FA-92CA85CD Contact: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 0: ACK sip:polycom_i1@10.0.5.133 SIP/2.0 (37) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK3dcc37b2;rport (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=as49e4c4e8 (52) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 3: To: "Line1" ;tag=A3D937FA-92CA85CD (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 5: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 6: CSeq: 102 ACK (13) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: SIP TIMER: Rescheduling retransmission #170 (1) INVITE - 5 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 88 ms (t1 44 ms (Retrans id #170)) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK7a91ba89;rport (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 4: CSeq: 103 INVITE (16) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 183 (19) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: o=- 1159700501 1159700502 IN IP4 10.0.5.130 (43) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: s=Polycom IP Phone (18) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: c=IN IP4 10.0.5.130 (19) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 101 (26) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Acked pending invite 103 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #170 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Stopping retransmission on '5fb12289210feb13773250987390be20@10.0.5.15' of Request 103: Match Found Oct 1 21:01:44 DEBUG[10546] chan_sip.c: SIP response 200 to RE-invite on outgoing call 5fb12289210feb13773250987390be20@10.0.5.15 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: build_route: Retaining previous route: Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 0: ACK sip:polycom_j1@10.0.5.130 SIP/2.0 (37) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6c5cf022;rport (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 6: CSeq: 103 ACK (13) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK7a91ba89;rport (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 4: CSeq: 103 INVITE (16) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 183 (19) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: o=- 1159700501 1159700502 IN IP4 10.0.5.130 (43) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: s=Polycom IP Phone (18) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: c=IN IP4 10.0.5.130 (19) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 101 (26) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Stopping retransmission on '5fb12289210feb13773250987390be20@10.0.5.15' of Request 103: Match Not Found Oct 1 21:01:44 DEBUG[10546] chan_sip.c: SIP response 200 to RE-invite on outgoing call 5fb12289210feb13773250987390be20@10.0.5.15 Oct 1 21:01:44 DEBUG[10546] chan_sip.c: build_route: Retaining previous route: Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 0: ACK sip:polycom_j1@10.0.5.130 SIP/2.0 (37) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK39372bcb;rport (60) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 6: CSeq: 103 ACK (13) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:44 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:49 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: REFER sip:0285661000@10.0.5.15 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bKa2b51be6E58FEDF From: ;tag=4BD59B6F-4FA1ECB6 To: "0285661000" ;tag=as64d4e590 CSeq: 2 REFER Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: REFER sip:0285661000@10.0.5.15 SIP/2.0 (38) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bKa2b51be6E58FEDF (57) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=4BD59B6F-4FA1ECB6 (55) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: "0285661000" ;tag=as64d4e590 (58) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: CSeq: 2 REFER (13) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: Refer-To: (143) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: Referred-By: (39) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 10: Max-Forwards: 70 (16) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 11: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 12: (0) Oct 1 21:01:49 VERBOSE[10546] logger.c: --- (12 headers 0 lines) --- Oct 1 21:01:49 DEBUG[10546] chan_sip.c: **** Received REFER (9) - Command in SIP REFER Oct 1 21:01:49 DEBUG[10546] chan_sip.c: SIP call transfer received for call 5cc79f12017802bd142be41c23692cab@10.0.5.15 (REFER)! Oct 1 21:01:49 VERBOSE[10546] logger.c: Transfer to 6109 in inside Oct 1 21:01:49 VERBOSE[10546] logger.c: Transfer from polycom_i1 in inside Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Assigning Replace-Call-ID Info f61336ce-c583d64c-8566eda7@10.0.5.133 to REPLACE_CALL_ID Oct 1 21:01:49 DEBUG[10546] chan_sip.c: 202 Accepted (supervised) Oct 1 21:01:49 DEBUG[10546] chan_agent.c: Asked for bridged channel on 'SIP/polycom_i1-081f9a60'/'Agent/6108', returning 'Zap/1-1' Oct 1 21:01:49 VERBOSE[10546] logger.c: -- Stopped music on hold on Zap/1-1 Oct 1 21:01:49 DEBUG[10546] channel.c: Set channel Zap/1-1 to write format alaw Oct 1 21:01:49 DEBUG[10546] channel.c: Scheduling timer at 0 sample intervals Oct 1 21:01:49 DEBUG[10546] chan_agent.c: Asked for bridged channel on 'Zap/1-1'/'Agent/6108', returning 'SIP/polycom_i1-081f9a60' Oct 1 21:01:49 DEBUG[10546] channel.c: Planning to masquerade channel Agent/6108 into the structure of SIP/polycom_i1-081d86d8 Oct 1 21:01:49 DEBUG[10546] channel.c: Done planning to masquerade channel Agent/6108 into the structure of SIP/polycom_i1-081d86d8 Oct 1 21:01:49 VERBOSE[10546] logger.c: Transmitting (no NAT) to 10.0.5.133:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bKa2b51be6E58FEDF;received=10.0.5.133 From: ;tag=4BD59B6F-4FA1ECB6 To: "0285661000" ;tag=as64d4e590 Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 202 Accepted (20) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bKa2b51be6E58FEDF;received=10.0.5.133 (77) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=4BD59B6F-4FA1ECB6 (55) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: "0285661000" ;tag=as64d4e590 (58) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: CSeq: 2 REFER (13) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: Contact: (35) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:49 VERBOSE[10546] logger.c: set_destination: Parsing for address/port to send to Oct 1 21:01:49 VERBOSE[10546] logger.c: set_destination: set destination to 10.0.5.133, port 5060 Oct 1 21:01:49 VERBOSE[10546] logger.c: Reliably Transmitting (no NAT) to 10.0.5.133:5060: NOTIFY sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0635dc49;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 Contact: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: NOTIFY sip:polycom_i1@10.0.5.133 SIP/2.0 (40) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0635dc49;rport (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: Contact: (35) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: CSeq: 103 NOTIFY (16) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: Event: refer;id=2 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 10: Subscription-state: terminated;reason=noresource (48) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 11: Content-Type: message/sipfrag;version=2.0 (41) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 12: Content-Length: 14 (18) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 13: (0) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #171 Oct 1 21:01:49 VERBOSE[10546] logger.c: set_destination: Parsing for address/port to send to Oct 1 21:01:49 VERBOSE[10546] logger.c: set_destination: set destination to 10.0.5.133, port 5060 Oct 1 21:01:49 VERBOSE[10546] logger.c: Reliably Transmitting (no NAT) to 10.0.5.133:5060: BYE sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0a343abb;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 Contact: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: BYE sip:polycom_i1@10.0.5.133 SIP/2.0 (37) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0a343abb;rport (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: Contact: (35) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: CSeq: 104 BYE (13) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #172 Oct 1 21:01:49 DEBUG[10580] channel.c: Actually Masquerading Agent/6108(6) into the structure of SIP/polycom_i1-081d86d8(6) Oct 1 21:01:49 DEBUG[10580] channel.c: Got clone lock for masquerade on 'Agent/6108' at 0x8247534 Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Hangup call Agent/6108, SIP callid f61336ce-c583d64c-8566eda7@10.0.5.133) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: update_call_counter(polycom_i1) - decrement call limit counter Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Updating call counter for outgoing call Oct 1 21:01:49 VERBOSE[10580] logger.c: Scheduling destruction of call 'f61336ce-c583d64c-8566eda7@10.0.5.133' in 32000 ms Oct 1 21:01:49 VERBOSE[10580] logger.c: set_destination: Parsing for address/port to send to Oct 1 21:01:49 VERBOSE[10580] logger.c: set_destination: set destination to 10.0.5.133, port 5060 Oct 1 21:01:49 VERBOSE[10580] logger.c: Reliably Transmitting (no NAT) to 10.0.5.133:5060: BYE sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK28b7c625;rport From: ;tag=as49e4c4e8 To: "Line1" ;tag=A3D937FA-92CA85CD Contact: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 0: BYE sip:polycom_i1@10.0.5.133 SIP/2.0 (37) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK28b7c625;rport (60) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 2: From: ;tag=as49e4c4e8 (52) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 3: To: "Line1" ;tag=A3D937FA-92CA85CD (60) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 5: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 6: CSeq: 103 BYE (13) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 10: (0) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #174 Oct 1 21:01:49 DEBUG[10580] channel.c: Putting channel Agent/6108 in 8/8 formats Oct 1 21:01:49 DEBUG[10580] channel.c: Released clone lock on 'SIP/polycom_i1-081d86d8' Oct 1 21:01:49 DEBUG[10566] channel.c: Didn't get a frame from channel: SIP/polycom_i1-081d86d8 Oct 1 21:01:49 DEBUG[10566] channel.c: Bridge stops bridging channels Zap/1-1 and SIP/polycom_i1-081d86d8 Oct 1 21:01:49 DEBUG[10566] channel.c: Hanging up zombie 'SIP/polycom_i1-081d86d8' Oct 1 21:01:49 DEBUG[10566] pbx.c: Spawn extension (open,s,9) exited non-zero on 'Zap/1-1' Oct 1 21:01:49 VERBOSE[10566] logger.c: == Spawn extension (open, s, 9) exited non-zero on 'Zap/1-1' Oct 1 21:01:49 DEBUG[10566] channel.c: Hanging up channel 'Zap/1-1' Oct 1 21:01:49 DEBUG[10566] chan_zap.c: zt_hangup(Zap/1-1) Oct 1 21:01:49 DEBUG[10566] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1 Oct 1 21:01:49 DEBUG[10566] chan_zap.c: Hangup: channel: 1 index = 0, normal = 14, callwait = -1, thirdcall = -1 Oct 1 21:01:49 DEBUG[10566] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call Oct 1 21:01:49 DEBUG[10566] chan_zap.c: disabled echo cancellation on channel 1 Oct 1 21:01:49 DEBUG[10566] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Oct 1 21:01:49 DEBUG[10566] chan_zap.c: Updated conferencing on 1, with 0 conference users Oct 1 21:01:49 DEBUG[10566] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 Oct 1 21:01:49 DEBUG[10566] chan_zap.c: disabled echo cancellation on channel 1 Oct 1 21:01:49 VERBOSE[10566] logger.c: -- Hungup 'Zap/1-1' Oct 1 21:01:49 DEBUG[10580] channel.c: Done Masquerading Agent/6108 (6) Oct 1 21:01:49 DEBUG[10580] rtp.c: Oooh, something is weird, backing out Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Sending reinvite on SIP '5fb12289210feb13773250987390be20@10.0.5.15' - It's audio soon redirected to IP 10.0.5.15 Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 0: INVITE sip:polycom_j1@10.0.5.130 SIP/2.0 (40) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6a244642;rport (60) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 6: CSeq: 104 INVITE (16) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 12: Content-Length: 210 (19) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 13: (0) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: v=0 (3) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: o=root 10580 10582 IN IP4 10.0.5.15 (35) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: s=session (9) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: m=audio 17486 RTP/AVP 8 101 (27) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 0: INVITE sip:polycom_j1@10.0.5.130 SIP/2.0 (40) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6a244642;rport (60) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 6: CSeq: 104 INVITE (16) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 9: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 10: X-asterisk-info: SIP re-invite (RTP bridge) (43) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 11: Content-Type: application/sdp (29) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 12: Content-Length: 210 (19) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 13: (0) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: v=0 (3) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: o=root 10580 10582 IN IP4 10.0.5.15 (35) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: s=session (9) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: c=IN IP4 10.0.5.15 (18) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: m=audio 17486 RTP/AVP 8 101 (27) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: a=fmtp:101 0-16 (15) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #175 Oct 1 21:01:49 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:49 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_i1 - state 2 (In use) Oct 1 21:01:49 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:49 DEBUG[10538] devicestate.c: Changing state for Zap/1 - state 0 (Unknown) Oct 1 21:01:49 DEBUG[10586] app_queue.c: Device 'SIP/polycom_i1' changed to state '2' (In use) but we don't care because they're not a member of any queue. Oct 1 21:01:49 DEBUG[10587] app_queue.c: Device 'Zap/1' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Oct 1 21:01:49 DEBUG[10546] chan_sip.c: SIP TIMER: Rescheduling retransmission #171 (1) NOTIFY - 4 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 84 ms (t1 42 ms (Retrans id #171)) Oct 1 21:01:49 VERBOSE[10546] logger.c: Retransmitting #1 (no NAT) to 10.0.5.133:5060: NOTIFY sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0635dc49;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 Contact: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Oct 1 21:01:49 DEBUG[10546] chan_sip.c: SIP TIMER: Rescheduling retransmission #172 (1) BYE - 8 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 84 ms (t1 42 ms (Retrans id #172)) Oct 1 21:01:49 VERBOSE[10546] logger.c: Retransmitting #1 (no NAT) to 10.0.5.133:5060: BYE sip:polycom_i1@10.0.5.133 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0a343abb;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 Contact: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Oct 1 21:01:49 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0635dc49;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 CSeq: 103 NOTIFY Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Content-Length: 0 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0635dc49;rport (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: CSeq: 103 NOTIFY (16) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: Event: refer;id=2 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:49 VERBOSE[10546] logger.c: --- (10 headers 0 lines) --- Oct 1 21:01:49 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #171 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Stopping retransmission on '5cc79f12017802bd142be41c23692cab@10.0.5.15' of Request 103: Match Found Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6a244642;rport (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: CSeq: 104 INVITE (16) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: Content-Type: application/sdp (29) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 10: Content-Length: 183 (19) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Line: v=0 (3) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Line: o=- 1159700501 1159700503 IN IP4 10.0.5.130 (43) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Line: s=Polycom IP Phone (18) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Line: c=IN IP4 10.0.5.130 (19) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Line: t=0 0 (5) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Line: m=audio 2226 RTP/AVP 8 101 (26) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Acked pending invite 104 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #175 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Stopping retransmission on '5fb12289210feb13773250987390be20@10.0.5.15' of Request 104: Match Found Oct 1 21:01:49 DEBUG[10546] chan_sip.c: SIP response 200 to RE-invite on outgoing call 5fb12289210feb13773250987390be20@10.0.5.15 Oct 1 21:01:49 DEBUG[10546] chan_agent.c: Asked for bridged channel on 'SIP/polycom_j1-081e0838'/'Agent/6108', returning 'SIP/polycom_i1-081f9a60' Oct 1 21:01:49 DEBUG[10546] chan_sip.c: build_route: Retaining previous route: Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: ACK sip:polycom_j1@10.0.5.130 SIP/2.0 (37) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK5e126b9b;rport (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: CSeq: 104 ACK (13) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:49 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: BYE sip:0285661000@10.0.5.15 SIP/2.0 Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bKd786b924549211C5 From: ;tag=4BD59B6F-4FA1ECB6 To: "0285661000" ;tag=as64d4e590 CSeq: 3 BYE Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Max-Forwards: 70 Content-Length: 0 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: BYE sip:0285661000@10.0.5.15 SIP/2.0 (36) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bKd786b924549211C5 (58) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=4BD59B6F-4FA1ECB6 (55) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: "0285661000" ;tag=as64d4e590 (58) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: CSeq: 3 BYE (11) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:49 VERBOSE[10546] logger.c: --- (10 headers 0 lines) --- Oct 1 21:01:49 DEBUG[10546] chan_sip.c: **** Received BYE (8) - Command in SIP BYE Oct 1 21:01:49 VERBOSE[10546] logger.c: Sending to 10.0.5.133 : 5060 (non-NAT) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Received bye, issuing owner hangup .Oct 1 21:01:49 VERBOSE[10546] logger.c: Transmitting (no NAT) to 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bKd786b924549211C5;received=10.0.5.133 From: ;tag=4BD59B6F-4FA1ECB6 To: "0285661000" ;tag=as64d4e590 Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.133;branch=z9hG4bKd786b924549211C5;received=10.0.5.133 (78) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=4BD59B6F-4FA1ECB6 (55) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: "0285661000" ;tag=as64d4e590 (58) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: CSeq: 3 BYE (11) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: Contact: (35) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 10: X-Asterisk-HangupCause: Normal Clearing (39) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 11: (0) Oct 1 21:01:49 DEBUG[10580] chan_agent.c: Bridge on 'SIP/polycom_i1-081f9a60' being cleared (2) Oct 1 21:01:49 DEBUG[10580] channel.c: Hanging up channel 'SIP/polycom_i1-081f9a60' Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Hangup call SIP/polycom_i1-081f9a60, SIP callid 5cc79f12017802bd142be41c23692cab@10.0.5.15) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: update_call_counter(polycom_i1) - decrement call limit counter Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Updating call counter for incoming call Oct 1 21:01:49 DEBUG[10580] channel.c: Didn't get a frame from channel: Agent/6108 Oct 1 21:01:49 DEBUG[10580] channel.c: Bridge stops bridging channels Agent/6108 and SIP/polycom_j1-081e0838 Oct 1 21:01:49 DEBUG[10580] channel.c: Hanging up channel 'SIP/polycom_j1-081e0838' Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Hangup call SIP/polycom_j1-081e0838, SIP callid 5fb12289210feb13773250987390be20@10.0.5.15) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: update_call_counter(polycom_j1) - decrement call limit counter Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Updating call counter for outgoing call Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 0: BYE sip:polycom_j1@10.0.5.130 SIP/2.0 (37) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6b49b4d8;rport (60) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 4: Contact: (29) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 6: CSeq: 105 BYE (13) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: Header 10: (0) Oct 1 21:01:49 DEBUG[10580] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #177 Oct 1 21:01:49 DEBUG[10580] app_dial.c: Exiting with DIALSTATUS=ANSWER. Oct 1 21:01:49 DEBUG[10580] app_macro.c: Spawn extension (macro-stdexten,s,28) exited non-zero on 'Agent/6108' in macro 'stdexten' Oct 1 21:01:49 VERBOSE[10580] logger.c: == Spawn extension (macro-stdexten, s, 28) exited non-zero on 'Agent/6108' in macro 'stdexten' Oct 1 21:01:49 DEBUG[10580] pbx.c: Spawn extension (macro-stdexten,s,28) exited non-zero on 'Agent/6108' Oct 1 21:01:49 VERBOSE[10580] logger.c: == Spawn extension (macro-stdexten, s, 28) exited non-zero on 'Agent/6108' Oct 1 21:01:49 DEBUG[10580] channel.c: Hanging up channel 'Agent/6108' Oct 1 21:01:49 DEBUG[10580] chan_agent.c: Hangup called for state Up Oct 1 21:01:49 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:49 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_i1 - state 1 (Not in use) Oct 1 21:01:49 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_i1 Oct 1 21:01:49 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_j1 Oct 1 21:01:49 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_j1 - state 1 (Not in use) Oct 1 21:01:49 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_j1 Oct 1 21:01:49 DEBUG[10538] devicestate.c: Changing state for Agent/6108 - state 1 (Not in use) Oct 1 21:01:49 DEBUG[10538] devicestate.c: Changing state for Agent/6108 - state 1 (Not in use) Oct 1 21:01:49 DEBUG[10588] app_queue.c: Device 'SIP/polycom_i1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Oct 1 21:01:49 DEBUG[10589] app_queue.c: Device 'SIP/polycom_j1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Oct 1 21:01:49 DEBUG[10590] app_queue.c: Device 'Agent/6108' changed to state '1' (Not in use) Oct 1 21:01:49 DEBUG[10591] app_queue.c: Device 'Agent/6108' changed to state '1' (Not in use) Oct 1 21:01:49 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0a343abb;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 CSeq: 104 BYE Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Content-Length: 0 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 500 Internal Server Error (33) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0a343abb;rport (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 4: CSeq: 104 BYE (13) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 8: Content-Length: 0 (17) Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Header 9: (0) Oct 1 21:01:49 VERBOSE[10546] logger.c: --- (9 headers 0 lines) --- Oct 1 21:01:49 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #172 Oct 1 21:01:49 DEBUG[10546] chan_sip.c: Stopping retransmission on '5cc79f12017802bd142be41c23692cab@10.0.5.15' of Request 104: Match Found Oct 1 21:01:49 VERBOSE[10546] logger.c: -- Incoming call: Got SIP response 500 "Internal Server Error" back from 10.0.5.133 Oct 1 21:01:49 VERBOSE[10546] logger.c: Destroying call '5cc79f12017802bd142be41c23692cab@10.0.5.15' Oct 1 21:01:50 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK28b7c625;rport From: ;tag=as49e4c4e8 To: "Line1" ;tag=A3D937FA-92CA85CD CSeq: 103 BYE Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Content-Length: 0 Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK28b7c625;rport (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=as49e4c4e8 (52) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 3: To: "Line1" ;tag=A3D937FA-92CA85CD (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 4: CSeq: 103 BYE (13) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 5: Call-ID: f61336ce-c583d64c-8566eda7@10.0.5.133 (46) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 8: Content-Length: 0 (17) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 9: (0) Oct 1 21:01:50 VERBOSE[10546] logger.c: --- (9 headers 0 lines) --- Oct 1 21:01:50 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #174 Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Stopping retransmission on 'f61336ce-c583d64c-8566eda7@10.0.5.133' of Request 103: Match Found Oct 1 21:01:50 VERBOSE[10546] logger.c: Destroying call 'f61336ce-c583d64c-8566eda7@10.0.5.133' Oct 1 21:01:50 DEBUG[10546] chan_sip.c: SIP TIMER: Rescheduling retransmission #177 (1) BYE - 8 Oct 1 21:01:50 DEBUG[10546] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 88 ms (t1 44 ms (Retrans id #177)) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6b49b4d8;rport (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 4: CSeq: 105 BYE (13) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 8: Content-Length: 0 (17) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 9: (0) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #177 Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Stopping retransmission on '5fb12289210feb13773250987390be20@10.0.5.15' of Request 105: Match Found Oct 1 21:01:50 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0635dc49;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 CSeq: 103 NOTIFY Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Content-Length: 0 Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0635dc49;rport (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 4: CSeq: 103 NOTIFY (16) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 7: Event: refer;id=2 (17) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 8: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 10: (0) Oct 1 21:01:50 VERBOSE[10546] logger.c: --- (10 headers 0 lines) --- Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for 5cc79f12017802bd142be41c23692cab@10.0.5.15 - SIP/2.0 (No RTP) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: That's odd... Got a response on a call we dont know about. Cseq 103 Cmd SIP/2.0 Oct 1 21:01:50 VERBOSE[10546] logger.c: Destroying call '5cc79f12017802bd142be41c23692cab@10.0.5.15' Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK6b49b4d8;rport (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 2: From: "Telephonist" ;tag=as25c07049 (55) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=6EF7C10-90ACD5F5 (52) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 4: CSeq: 105 BYE (13) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5fb12289210feb13773250987390be20@10.0.5.15 (51) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 8: Content-Length: 0 (17) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 9: (0) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for 5fb12289210feb13773250987390be20@10.0.5.15 - SIP/2.0 (No RTP) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: That's odd... Got a response on a call we dont know about. Cseq 105 Cmd SIP/2.0 Oct 1 21:01:50 VERBOSE[10546] logger.c: <-- SIP read from 10.0.5.133:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0a343abb;rport From: "0285661000" ;tag=as64d4e590 To: ;tag=4BD59B6F-4FA1ECB6 CSeq: 104 BYE Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 Contact: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 Content-Length: 0 Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 500 Internal Server Error (33) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK0a343abb;rport (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 2: From: "0285661000" ;tag=as64d4e590 (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 3: To: ;tag=4BD59B6F-4FA1ECB6 (53) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 4: CSeq: 104 BYE (13) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 5cc79f12017802bd142be41c23692cab@10.0.5.15 (51) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 8: Content-Length: 0 (17) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 9: (0) Oct 1 21:01:50 VERBOSE[10546] logger.c: --- (9 headers 0 lines) --- Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Allocating new SIP dialog for 5cc79f12017802bd142be41c23692cab@10.0.5.15 - SIP/2.0 (No RTP) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: That's odd... Got a response on a call we dont know about. Cseq 104 Cmd SIP/2.0 Oct 1 21:01:50 VERBOSE[10546] logger.c: Destroying call '5cc79f12017802bd142be41c23692cab@10.0.5.15' Oct 1 21:01:50 VERBOSE[10543] logger.c: -- Channel 0/10, span 1 got hangup request Oct 1 21:01:50 DEBUG[10562] channel.c: Didn't get a frame from channel: Zap/10-1 Oct 1 21:01:50 DEBUG[10562] channel.c: Bridge stops bridging channels SIP/polycom_p1-081fc808 and Zap/10-1 Oct 1 21:01:50 DEBUG[10562] channel.c: Hanging up channel 'Zap/10-1' Oct 1 21:01:50 DEBUG[10562] chan_zap.c: zt_hangup(Zap/10-1) Oct 1 21:01:50 DEBUG[10562] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/10-1 Oct 1 21:01:50 DEBUG[10562] chan_zap.c: Hangup: channel: 10 index = 0, normal = 23, callwait = -1, thirdcall = -1 Oct 1 21:01:50 DEBUG[10562] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call Oct 1 21:01:50 DEBUG[10562] chan_zap.c: disabled echo cancellation on channel 10 Oct 1 21:01:50 DEBUG[10562] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/10-1 Oct 1 21:01:50 DEBUG[10562] chan_zap.c: Updated conferencing on 10, with 0 conference users Oct 1 21:01:50 DEBUG[10562] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/10-1 Oct 1 21:01:50 DEBUG[10562] chan_zap.c: disabled echo cancellation on channel 10 Oct 1 21:01:50 VERBOSE[10562] logger.c: -- Hungup 'Zap/10-1' Oct 1 21:01:50 DEBUG[10562] app_dial.c: Exiting with DIALSTATUS=ANSWER. Oct 1 21:01:50 DEBUG[10562] app_macro.c: Spawn extension (macro-dialout,s,5) exited non-zero on 'SIP/polycom_p1-081fc808' in macro 'dialout' Oct 1 21:01:50 VERBOSE[10562] logger.c: == Spawn extension (macro-dialout, s, 5) exited non-zero on 'SIP/polycom_p1-081fc808' in macro 'dialout' Oct 1 21:01:50 DEBUG[10562] pbx.c: Spawn extension (macro-dialout,s,5) exited non-zero on 'SIP/polycom_p1-081fc808' Oct 1 21:01:50 VERBOSE[10562] logger.c: == Spawn extension (macro-dialout, s, 5) exited non-zero on 'SIP/polycom_p1-081fc808' Oct 1 21:01:50 DEBUG[10562] channel.c: Hanging up channel 'SIP/polycom_p1-081fc808' Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Hangup call SIP/polycom_p1-081fc808, SIP callid 60f08dfd-b4a60c93-550fc47c@10.0.5.129) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: update_call_counter(polycom_p1) - decrement call limit counter Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Updating call counter for incoming call Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 0: BYE sip:polycom_p1@10.0.5.129 SIP/2.0 (37) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK55e83c68;rport (60) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 2: From: ;tag=as3041f41c (57) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 3: To: "Line1" ;tag=177D7E89-F26F3C02 (60) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 4: Contact: (34) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 5: Call-ID: 60f08dfd-b4a60c93-550fc47c@10.0.5.129 (46) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 6: CSeq: 102 BYE (13) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 8: Max-Forwards: 70 (16) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 9: Content-Length: 0 (17) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: Header 10: (0) Oct 1 21:01:50 DEBUG[10562] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #179 Oct 1 21:01:50 DEBUG[10538] devicestate.c: Changing state for Zap/10 - state 0 (Unknown) Oct 1 21:01:50 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_p1 Oct 1 21:01:50 DEBUG[10538] devicestate.c: Changing state for SIP/polycom_p1 - state 1 (Not in use) Oct 1 21:01:50 DEBUG[10538] chan_sip.c: Checking device state for peer polycom_p1 Oct 1 21:01:50 DEBUG[10592] app_queue.c: Device 'Zap/10' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. Oct 1 21:01:50 DEBUG[10593] app_queue.c: Device 'SIP/polycom_p1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 0: SIP/2.0 200 OK (14) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.0.5.15:5060;branch=z9hG4bK55e83c68;rport (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 2: From: ;tag=as3041f41c (57) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 3: To: "Line1" ;tag=177D7E89-F26F3C02 (60) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 4: CSeq: 102 BYE (13) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 5: Call-ID: 60f08dfd-b4a60c93-550fc47c@10.0.5.129 (46) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 6: Contact: (36) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 7: User-Agent: PolycomSoundPointIP-SPIP_300-UA/1.6.7.0098 (54) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 8: Content-Length: 0 (17) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Header 9: (0) Oct 1 21:01:50 DEBUG[10546] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #179 Oct 1 21:01:50 DEBUG[10546] chan_sip.c: Stopping retransmission on '60f08dfd-b4a60c93-550fc47c@10.0.5.129' of Request 102: Match Found Oct 1 21:02:07 VERBOSE[10561] logger.c: Beginning asterisk shutdown.... Oct 1 21:02:07 VERBOSE[10561] logger.c: Executing last minute cleanups Oct 1 21:02:07 VERBOSE[10561] logger.c: == Destroying musiconhold processes Oct 1 21:02:07 VERBOSE[10561] logger.c: Asterisk cleanly ending (0). Oct 1 21:02:07 DEBUG[10561] asterisk.c: Asterisk ending (0).