-= Registered Asterisk Dial Plan Hints =- 8716@private-extensions : SIP/8716 State:Unavailable Watchers 0 8715@private-extensions : SIP/8715 State:Unavailable Watchers 0 8714@private-extensions : SIP/8714 State:Unavailable Watchers 0 8713@private-extensions : SIP/8713 State:Unavailable Watchers 0 8712@private-extensions : SIP/8712 State:Unavailable Watchers 0 8711@private-extensions : SIP/8711 State:Unavailable Watchers 0 8710@private-extensions : SIP/8710 State:Unavailable Watchers 0 8696@private-extensions : SIP/8696 State:Unavailable Watchers 0 8695@private-extensions : SIP/8695 State:Unavailable Watchers 0 8694@private-extensions : SIP/8694 State:Unavailable Watchers 0 8693@private-extensions : SIP/8693 State:Unavailable Watchers 0 8692@private-extensions : SIP/8692 State:Unavailable Watchers 0 8691@private-extensions : SIP/8691 State:Unavailable Watchers 0 8690@private-extensions : SIP/8690 State:Unavailable Watchers 0 8169@private-extensions : SIP/8169 State:Unavailable Watchers 0 8168@private-extensions : SIP/8168 State:Unavailable Watchers 0 8166@private-extensions : SIP/8166 State:Unavailable Watchers 0 8162@private-extensions : SIP/8162 State:Unavailable Watchers 0 8161@private-extensions : SIP/8161 State:Unavailable Watchers 0 8160@private-extensions : SIP/8160 State:Unavailable Watchers 0 8159@private-extensions : SIP/8159 State:Unavailable Watchers 0 8158@private-extensions : SIP/8158 State:Unavailable Watchers 0 8157@private-extensions : SIP/8157 State:Unavailable Watchers 0 8156@private-extensions : SIP/8156 State:Unavailable Watchers 0 8155@private-extensions : SIP/8155 State:Unavailable Watchers 0 8154@private-extensions : SIP/8154 State:Unavailable Watchers 0 8153@private-extensions : SIP/8153 State:Unavailable Watchers 0 8152@private-extensions : SIP/8152 State:Unavailable Watchers 0 8151@private-extensions : SIP/8151 State:Unavailable Watchers 0 8150@private-extensions : SIP/8150 State:Unavailable Watchers 0 8016@private-extensions : SIP/8016 State:Unavailable Watchers 0 8015@private-extensions : SIP/8015 State:Unavailable Watchers 0 8014@private-extensions : SIP/8014 State:Unavailable Watchers 0 8013@private-extensions : SIP/8013 State:Unavailable Watchers 0 8012@private-extensions : SIP/8012 State:Unavailable Watchers 0 8011@private-extensions : SIP/8011 State:Unavailable Watchers 0 8010@private-extensions : SIP/8010 State:Unavailable Watchers 0 8009@private-extensions : SIP/8009 State:Unavailable Watchers 0 8008@private-extensions : SIP/8008 State:Unavailable Watchers 0 8007@private-extensions : SIP/8007 State:Unavailable Watchers 0 8006@private-extensions : SIP/8006 State:Unavailable Watchers 0 8005@private-extensions : SIP/8005 State:Unavailable Watchers 0 8004@private-extensions : SIP/8004 State:Unavailable Watchers 0 8003@private-extensions : SIP/8003 State:Unavailable Watchers 0 8002@private-extensions : SIP/8002 State:Unavailable Watchers 0 8001@private-extensions : SIP/8001 State:Unavailable Watchers 0 4012@private-extensions : SIP/4012 State:Idle Watchers 1 4011@private-extensions : SIP/4011 State:Idle Watchers 1 4010@private-extensions : SIP/4010 State:Idle Watchers 0 4009@private-extensions : SIP/4009 State:Unavailable Watchers 0 4008@private-extensions : SIP/4008 State:Unavailable Watchers 0 4007@private-extensions : SIP/4007 State:Unavailable Watchers 0 4006@private-extensions : SIP/4006 State:Unavailable Watchers 0 4005@private-extensions : SIP/4005 State:Unavailable Watchers 0 4004@private-extensions : SIP/4004 State:Unavailable Watchers 0 4003@private-extensions : SIP/4003 State:Unavailable Watchers 0 4002@private-extensions : SIP/4002 State:Unavailable Watchers 0 4001@private-extensions : SIP/4001 State:Unavailable Watchers 0 3844@public-extensions : SIP/3844 State:Unavailable Watchers 0 3842@public-extensions : SIP/3842 State:Unavailable Watchers 0 3838@public-extensions : SIP/3838 State:Unavailable Watchers 0 3727@public-extensions : SIP/3727 State:Unavailable Watchers 0 3712@public-extensions : SIP/3712 State:Unavailable Watchers 0 3711@public-extensions : SIP/3711 State:Unavailable Watchers 0 3700@public-extensions : SIP/3700 State:Unavailable Watchers 0 3690@public-extensions : SIP/3690 State:Unavailable Watchers 0 3681@public-extensions : SIP/3681 State:Unavailable Watchers 0 3675@public-extensions : SIP/3675 State:Unavailable Watchers 0 3674@public-extensions : SIP/3674 State:Unavailable Watchers 0 3673@public-extensions : SIP/3673 State:Unavailable Watchers 0 3672@public-extensions : SIP/3672 State:Unavailable Watchers 0 3671@public-extensions : SIP/3671 State:Unavailable Watchers 0 3670@public-extensions : SIP/3670 State:Unavailable Watchers 0 3668@public-extensions : SIP/3668 State:Unavailable Watchers 0 3667@public-extensions : SIP/3667 State:Unavailable Watchers 0 3665@public-extensions : SIP/3665 State:Unavailable Watchers 0 3664@public-extensions : SIP/3664 State:Unavailable Watchers 0 3663@public-extensions : SIP/3663 State:Unavailable Watchers 0 3662@public-extensions : SIP/3662 State:Unavailable Watchers 0 3661@public-extensions : SIP/3661 State:Unavailable Watchers 0 3660@public-extensions : SIP/3660 State:Unavailable Watchers 0 3658@public-extensions : SIP/3658 State:Unavailable Watchers 0 3655@public-extensions : SIP/3655 State:Unavailable Watchers 0 3654@public-extensions : SIP/3654 State:Unavailable Watchers 0 3653@public-extensions : SIP/3653 State:Unavailable Watchers 0 3652@public-extensions : SIP/3652 State:Unavailable Watchers 0 3651@public-extensions : SIP/3651 State:Unavailable Watchers 0 3031@public-extensions : SIP/3031 State:Unavailable Watchers 0 3030@public-extensions : SIP/3030 State:Unavailable Watchers 0 2499@public-extensions : SIP/2499 State:Unavailable Watchers 0 2488@public-extensions : SIP/2488 State:Unavailable Watchers 0 2485@public-extensions : SIP/2485 State:Unavailable Watchers 0 2478@public-extensions : SIP/2478 State:Unavailable Watchers 0 2472@public-extensions : SIP/2472 State:Unavailable Watchers 0 2471@public-extensions : SIP/2471 State:Idle Watchers 0 2470@public-extensions : SIP/2470 State:Unavailable Watchers 0 2469@public-extensions : SIP/2469 State:Unavailable Watchers 0 2468@public-extensions : SIP/2468 State:Unavailable Watchers 0 2457@public-extensions : SIP/2457 State:Unavailable Watchers 0 2456@public-extensions : SIP/2456 State:Unavailable Watchers 0 2448@public-extensions : SIP/2448 State:Unavailable Watchers 0 2440@public-extensions : SIP/2440 State:Unavailable Watchers 0 2439@public-extensions : SIP/2439 State:Unavailable Watchers 0 2438@public-extensions : SIP/2438 State:Unavailable Watchers 0 2426@public-extensions : SIP/2426 State:Unavailable Watchers 0 2420@public-extensions : SIP/2420 State:Idle Watchers 0 2412@public-extensions : SIP/2412 State:Unavailable Watchers 0 2397@public-extensions : SIP/2397 State:Unavailable Watchers 0 2390@public-extensions : SIP/2390 State:Unavailable Watchers 0 2382@public-extensions : SIP/2382 State:Unavailable Watchers 0 2381@public-extensions : SIP/2381 State:Unavailable Watchers 0 2380@public-extensions : SIP/2380 State:Unavailable Watchers 0 2372@public-extensions : SIP/2372 State:Unavailable Watchers 0 2348@public-extensions : SIP/2348-Polycom State:Unavailable Watchers 0 2331@public-extensions : SIP/2331 State:Unavailable Watchers 0 2323@public-extensions : SIP/2323 State:Unavailable Watchers 0 2313@public-extensions : SIP/2313 State:Unavailable Watchers 0 2312@public-extensions : SIP/2312 State:Idle Watchers 0 2309@public-extensions : SIP/2309 State:Unavailable Watchers 0 2308@public-extensions : SIP/2308 State:Unavailable Watchers 0 2306@public-extensions : SIP/2306-Polycom&SIP State:Unavailable Watchers 0 2305@public-extensions : SIP/2305 State:Unavailable Watchers 0 2304@public-extensions : SIP/2304 State:Unavailable Watchers 0 2303@public-extensions : SIP/2303 State:Unavailable Watchers 0 2301@public-extensions : SIP/2301 State:Unavailable Watchers 0 2295@public-extensions : SIP/2295 State:Unavailable Watchers 0 2290@public-extensions : SIP/2290-ridgej State:Unavailable Watchers 0 2266@public-extensions : SIP/2266 State:Unavailable Watchers 0 2238@public-extensions : SIP/2238 State:Unavailable Watchers 0 2235@public-extensions : SIP/2235 State:Unavailable Watchers 0 2230@public-extensions : SIP/2230 State:Unavailable Watchers 0 2229@public-extensions : SIP/2229 State:Unavailable Watchers 0 2228@public-extensions : SIP/2228 State:Unavailable Watchers 0 2226@public-extensions : SIP/2226 State:Unavailable Watchers 0 2225@public-extensions : SIP/2225 State:Unavailable Watchers 0 2218@public-extensions : SIP/2218 State:Unavailable Watchers 0 2209@public-extensions : SIP/2206-mitchells&S State:Unavailable Watchers 0 2206@public-extensions : SIP/2206-ridgej State:Unavailable Watchers 0 ---------------- - 138 hints registered *CLI> show hints[Feb 8 12:29:50] DEBUG[12435]: chan_iax2.c:4775 raw_hangup: Raw Hangup 204.239.10.89:1627, src=0, dst=0 ip show inuse * User name In use Limit 4012 0 200 4011 0 200 4010 0 200 2348-Polycom 0 200 * Peer name In use Limit 4012 0/0 200 4011 0/0 200 4010 0/0 200 2348-Polycom 0/0 200 *CLI> *CLI> *CLI> [Feb 8 12:29:55] DEBUG[12438]: chan_sip.c:2009 __sip_autodestruct: Auto destroying SIP dialog '1f835161-a09fffd7-163480ca@172.16.16.135' [Feb 8 12:29:55] DEBUG[12438]: chan_sip.c:3108 sip_destroy: Destroying SIP dialog 1f835161-a09fffd7-163480ca@172.16.16.135 [Feb 8 12:29:55] Really destroying SIP dialog '1f835161-a09fffd7-163480ca@172.16.16.135' Method: REGISTER [Feb 8 12:30:02] <--- SIP read from 172.16.16.101:5060 ---> INVITE sip:4012@voip.dogmatix.dnv.org;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5ecc4ed471733539 From: "4011" ;tag=225C83B2-C4872DCD To: CSeq: 1 INVITE Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1170966594 1170966594 IN IP4 172.16.16.101 s=Polycom IP Phone c=IN IP4 172.16.16.101 t=0 0 m=audio 2238 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: INVITE sip:4012@voip.dogmatix.dnv.org;user=phone SIP/2.0 (56) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5ecc4ed471733539 (66) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=225C83B2-C4872DCD (67) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (47) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1 INVITE (14) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Supported: 100rel,replaces (26) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow-Events: talk,hold,conference (34) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Max-Forwards: 70 (16) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 13: Content-Length: 239 (19) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 14: (0) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: v=0 (3) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: o=- 1170966594 1170966594 IN IP4 172.16.16.101 (46) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: s=Polycom IP Phone (18) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: c=IN IP4 172.16.16.101 (22) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: t=0 0 (5) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: m=audio 2238 RTP/AVP 0 8 18 101 (31) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 8 12:30:02] --- (14 headers 10 lines) --- [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2574 do_setnat: Setting NAT on RTP to Off [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for 5588e51e-c415a168-c97a9e83@172.16.16.101 - INVITE (With RTP) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:1679 parse_sip_options: Begin: parsing SIP "Supported: 100rel,replaces" [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:1687 parse_sip_options: Found SIP option: -100rel- [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:1693 parse_sip_options: Matched SIP option: 100rel [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:1687 parse_sip_options: Found SIP option: -replaces- [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:1693 parse_sip_options: Matched SIP option: replaces [Feb 8 12:30:02] Sending to 172.16.16.101 : 5060 (no NAT) [Feb 8 12:30:02] Using INVITE request as basis request - 5588e51e-c415a168-c97a9e83@172.16.16.101 [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2574 do_setnat: Setting NAT on RTP to Off [Feb 8 12:30:02] <--- Reliably Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5ecc4ed471733539;received=172.16.16.101 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as7ea06bfe Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1269c090" Content-Length: 0 <------------> [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #426 [Feb 8 12:30:02] Scheduling destruction of SIP dialog '5588e51e-c415a168-c97a9e83@172.16.16.101' in 32000 ms (Method: INVITE) [Feb 8 12:30:02] Found user '4011' [Feb 8 12:30:02] <--- SIP read from 172.16.16.101:5060 ---> ACK sip:4012@voip.dogmatix.dnv.org SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5ecc4ed471733539 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as7ea06bfe CSeq: 1 ACK Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Max-Forwards: 70 Content-Length: 0 <-------------> [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: ACK sip:4012@voip.dogmatix.dnv.org SIP/2.0 (42) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5ecc4ed471733539 (66) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=225C83B2-C4872DCD (67) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=as7ea06bfe (62) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1 ACK (11) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Content-Length: 0 (17) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: (0) [Feb 8 12:30:02] --- (11 headers 0 lines) --- [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #426 [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '5588e51e-c415a168-c97a9e83@172.16.16.101' of Response 1: Match Not Found [Feb 8 12:30:02] <--- SIP read from 172.16.16.101:5060 ---> INVITE sip:4012@voip.dogmatix.dnv.org;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5d82e3176E478BFC From: "4011" ;tag=225C83B2-C4872DCD To: CSeq: 2 INVITE Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="a34a604bd4f802b2aa2258171262e82c", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1170966594 1170966594 IN IP4 172.16.16.101 s=Polycom IP Phone c=IN IP4 172.16.16.101 t=0 0 m=audio 2238 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: INVITE sip:4012@voip.dogmatix.dnv.org;user=phone SIP/2.0 (56) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5d82e3176E478BFC (66) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=225C83B2-C4872DCD (67) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (47) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 2 INVITE (14) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Supported: 100rel,replaces (26) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow-Events: talk,hold,conference (34) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="a34a604bd4f802b2aa2258171262e82c", algorithm=MD5 (188) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Max-Forwards: 70 (16) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 13: Content-Type: application/sdp (29) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 14: Content-Length: 239 (19) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 15: (0) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: v=0 (3) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: o=- 1170966594 1170966594 IN IP4 172.16.16.101 (46) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: s=Polycom IP Phone (18) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: c=IN IP4 172.16.16.101 (22) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: t=0 0 (5) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: m=audio 2238 RTP/AVP 0 8 18 101 (31) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 8 12:30:02] --- (15 headers 10 lines) --- [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 8 12:30:02] Sending to 172.16.16.101 : 5060 (no NAT) [Feb 8 12:30:02] Using INVITE request as basis request - 5588e51e-c415a168-c97a9e83@172.16.16.101 [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2574 do_setnat: Setting NAT on RTP to Off [Feb 8 12:30:02] Found user '4011' [Feb 8 12:30:02] Found RTP audio format 0 [Feb 8 12:30:02] Found RTP audio format 8 [Feb 8 12:30:02] Found RTP audio format 18 [Feb 8 12:30:02] Found RTP audio format 101 [Feb 8 12:30:02] Peer audio RTP is at port 172.16.16.101:2238 [Feb 8 12:30:02] Found description format PCMU for ID 0 [Feb 8 12:30:02] Found description format PCMA for ID 8 [Feb 8 12:30:02] Found description format G729 for ID 18 [Feb 8 12:30:02] Found description format telephone-event for ID 101 [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:5105 process_sdp: T38 state changed to 0 on channel [Feb 8 12:30:02] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 8 12:30:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 8 12:30:02] Peer audio RTP is at port 172.16.16.101:2238 [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:5185 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:13363 handle_request_invite: Checking SIP call limits for device 4011 [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:3002 update_call_counter: Updating call counter for incoming call [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:3076 update_call_counter: Call from peer '4011' is 1 out of 200 [Feb 8 12:30:02] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011 [Feb 8 12:30:02] Looking for 4012 in internal (domain voip.dogmatix.dnv.org) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:3802 sip_new: *** Our native formats are 0x4 (ulaw) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:3803 sip_new: *** Joint capabilities are 0x4 (ulaw) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:3804 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:3805 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:3828 sip_new: This channel will not be able to handle video. [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:7953 build_route: build_route: Contact hop: [Feb 8 12:30:02] list_route: hop: [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:13438 handle_request_invite: SIP/4011-0897d348: New call is still down.... Trying... [Feb 8 12:30:02] <--- Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5d82e3176E478BFC;received=172.16.16.101 From: "4011" ;tag=225C83B2-C4872DCD To: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:02] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011-0897d348 [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 2 (In use) [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:02] Reliably Transmitting (no NAT) to 172.16.16.135:5060: NOTIFY sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3b4fd64d;rport From: ;tag=as1a9ebeca To: "4010" ;tag=D2C77ADC-9D78A5CB Contact: Call-ID: e9654c50-a338ad2e-417f8575@172.16.16.135 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 367
--- [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #428 [Feb 8 12:30:02] Extension Changed 4011 new state InUse for Notify User 4010 [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 2 (In use) [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'Goto' [Feb 8 12:30:02] -- Executing [4012@internal:1] Goto("SIP/4011-0897d348", "ITS|4012|1") in new stack [Feb 8 12:30:02] -- Goto (ITS,4012,1) [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '4011' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'Macro' [Feb 8 12:30:02] -- Executing [4012@ITS:1] Macro("SIP/4011-0897d348", "dnv-incoming-call-handler|4011|4012") in new stack [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is 'ITS Test 3' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'NoOp' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:1] NoOp("SIP/4011-0897d348", "ITS Test 3") in new stack [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is 'ITS Test 3' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:2] GotoIf("SIP/4011-0897d348", "0?SetVoiceMailContext") in new stack [Feb 8 12:30:02] DEBUG[12501]: pbx.c:5947 pbx_builtin_gotoif: Not taking any branch [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is 'SIP' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '1' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:3] GotoIf("SIP/4011-0897d348", "1?:SetVoiceMailContext") in new stack [Feb 8 12:30:02] DEBUG[12501]: pbx.c:5947 pbx_builtin_gotoif: Not taking any branch [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is 'ITS Test 3' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'SIPCalledRPID' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:4] SIPCalledRPID("SIP/4011-0897d348", "ITS Test 3|4012") in new stack [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:5] GotoIf("SIP/4011-0897d348", "0?:NVDPL") in new stack [Feb 8 12:30:02] -- Goto (macro-dnv-incoming-call-handler,s,8) [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:8] GotoIf("SIP/4011-0897d348", "0?:Default") in new stack [Feb 8 12:30:02] -- Goto (macro-dnv-incoming-call-handler,s,12) [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'Set' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:12] Set("SIP/4011-0897d348", "vm_context=default") in new stack [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '1' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '1' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:13] GotoIf("SIP/4011-0897d348", "0?:Proceed") in new stack [Feb 8 12:30:02] -- Goto (macro-dnv-incoming-call-handler,s,19) [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '"ITS Test 2" <4011>' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'NoOp' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:19] NoOp("SIP/4011-0897d348", ""ITS Test 2" <4011>") in new stack [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '4011' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:20] GotoIf("SIP/4011-0897d348", "0?:IdentifyCID") in new stack [Feb 8 12:30:02] -- Goto (macro-dnv-incoming-call-handler,s,24) [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '4011' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:24] GotoIf("SIP/4011-0897d348", "0?:Finish") in new stack [Feb 8 12:30:02] -- Goto (macro-dnv-incoming-call-handler,s,26) [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '"ITS Test 2" <4011>' [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'NoOp' [Feb 8 12:30:02] -- Executing [s@macro-dnv-incoming-call-handler:26] NoOp("SIP/4011-0897d348", ""ITS Test 2" <4011>") in new stack [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'Macro' [Feb 8 12:30:02] -- Executing [4012@ITS:2] Macro("SIP/4011-0897d348", "dnv-standard-dialer|SIP/4012|20|o") in new stack [Feb 8 12:30:02] DEBUG[12501]: pbx.c:1769 pbx_extension_helper: Launching 'Dial' [Feb 8 12:30:02] -- Executing [s@macro-dnv-standard-dialer:1] Dial("SIP/4011-0897d348", "SIP/4012|20|o") in new stack [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:15246 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:2574 do_setnat: Setting NAT on RTP to Off [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:3802 sip_new: *** Our native formats are 0x4 (ulaw) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:3803 sip_new: *** Joint capabilities are 0x0 (nothing) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:3804 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:3805 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:3807 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:3828 sip_new: This channel will not be able to handle video. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-standard-dialer-s-1. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable ARG3. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable ARG2. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable ARG1. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable MACRO_PRIORITY. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable MACRO_CONTEXT. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable MACRO_EXTEN. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-ITS-4012-2. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-26. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-24. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-20. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-19. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-13. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable vm_context. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-12. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-8. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-5. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-4. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-3. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-2. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-1. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-ITS-4012-1. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-internal-4012-1. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Feb 8 12:30:02] DEBUG[12501]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPURI. [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:2829 sip_call: Outgoing Call for 4012 [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:3002 update_call_counter: Updating call counter for outgoing call [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:3076 update_call_counter: Call to peer '4012' is 1 out of 200 [Feb 8 12:30:02] DEBUG[12501]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4012 [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4012 - state 6 (Ringing) [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:02] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:02] Reliably Transmitting (no NAT) to 172.16.16.135:5060: NOTIFY sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1a0a0080;rport From: ;tag=as7fcef0ca To: "4010" ;tag=B5A00269-C426A3B4 Contact: Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 367
--- [Feb 8 12:30:02] DEBUG[12424]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #429 [Feb 8 12:30:02] Extension Changed 4012 new state Ringing for Notify User 4010 [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:2844 sip_call: Our T38 capability (0), joint T38 capability (0) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:6178 add_sdp: ** Our capability: 0x6 (gsm|ulaw) Video flag: False [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:6179 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Feb 8 12:30:02] Audio is at 172.16.16.12 port 13936 [Feb 8 12:30:02] Adding codec 0x4 (ulaw) to SDP [Feb 8 12:30:02] Adding codec 0x2 (gsm) to SDP [Feb 8 12:30:02] Adding non-codec 0x1 (telephone-event) to SDP [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:6310 add_sdp: -- Done with adding codecs to SDP [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:6355 add_sdp: Done building SDP. Settling with this capability: 0x6 (gsm|ulaw) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 0: INVITE sip:4012@172.16.16.133:5060 SIP/2.0 (42) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK46db74e2;rport (63) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 2: From: "ITS Test 2" ;tag=as054a4c33 (57) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 4: Contact: (32) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 (54) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:02 GMT (35) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 13: Content-Length: 263 (19) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4562 parse_request: Header 14: (0) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: v=0 (3) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: o=root 12420 12420 IN IP4 172.16.16.12 (38) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: s=session (9) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: c=IN IP4 172.16.16.12 (21) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: t=0 0 (5) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: m=audio 13936 RTP/AVP 0 3 101 (29) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: a=ptime:20 (10) [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:4594 parse_request: Line: a=sendrecv (10) [Feb 8 12:30:02] Reliably Transmitting (no NAT) to 172.16.16.133:5060: INVITE sip:4012@172.16.16.133:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK46db74e2;rport From: "ITS Test 2" ;tag=as054a4c33 To: Contact: Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=root 12420 12420 IN IP4 172.16.16.12 s=session c=IN IP4 172.16.16.12 t=0 0 m=audio 13936 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 8 12:30:02] DEBUG[12501]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #430 [Feb 8 12:30:02] -- Called 4012 [Feb 8 12:30:02] DEBUG[12502]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:02] DEBUG[12503]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:02] DEBUG[12504]: app_queue.c:546 changethread: Device 'SIP/4012' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Feb 8 12:30:02] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3b4fd64d;rport From: ;tag=as1a9ebeca To: "4010" ;tag=D2C77ADC-9D78A5CB CSeq: 104 NOTIFY Call-ID: e9654c50-a338ad2e-417f8575@172.16.16.135 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3b4fd64d;rport (63) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=as1a9ebeca (53) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "4010" ;tag=D2C77ADC-9D78A5CB (65) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 104 NOTIFY (16) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: e9654c50-a338ad2e-417f8575@172.16.16.135 (49) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: presence (15) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:02] --- (10 headers 0 lines) --- [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #428 [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on 'e9654c50-a338ad2e-417f8575@172.16.16.135' of Request 104: Match Not Found [Feb 8 12:30:02] SIP Response message for INCOMING dialog NOTIFY arrived [Feb 8 12:30:02] <--- SIP read from 172.16.16.133:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK46db74e2;rport From: "ITS Test 2" ;tag=as054a4c33 To: ;tag=C207580-50CC03A7 CSeq: 102 INVITE Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 100 Trying (18) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK46db74e2;rport (63) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "ITS Test 2" ;tag=as054a4c33 (57) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=C207580-50CC03A7 (54) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 INVITE (16) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 (54) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Content-Length: 0 (17) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: (0) [Feb 8 12:30:02] --- (9 headers 0 lines) --- [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2121 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #430 - INVITE (got response) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2130 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7b13f17e349ea65f031be7f1482226e2@172.16.16.12' Request 102: Found [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:11610 handle_response_invite: SIP response 100 to standard invite [Feb 8 12:30:02] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1a0a0080;rport From: ;tag=as7fcef0ca To: "4010" ;tag=B5A00269-C426A3B4 CSeq: 103 NOTIFY Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1a0a0080;rport (63) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=as7fcef0ca (53) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "4010" ;tag=B5A00269-C426A3B4 (65) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 103 NOTIFY (16) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 (49) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: presence (15) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:02] --- (10 headers 0 lines) --- [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #429 [Feb 8 12:30:02] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on 'ae75b12d-d9314ac3-30e8f7c6@172.16.16.135' of Request 103: Match Not Found [Feb 8 12:30:02] SIP Response message for INCOMING dialog NOTIFY arrived [Feb 8 12:30:03] <--- SIP read from 172.16.16.133:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK46db74e2;rport From: "ITS Test 2" ;tag=as054a4c33 To: ;tag=C207580-50CC03A7 CSeq: 102 INVITE Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK46db74e2;rport (63) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "ITS Test 2" ;tag=as054a4c33 (57) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=C207580-50CC03A7 (54) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 INVITE (16) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 (54) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Allow-Events: talk,hold,conference (34) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:03] --- (10 headers 0 lines) --- [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:2130 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '7b13f17e349ea65f031be7f1482226e2@172.16.16.12' Request 102: Found [Feb 8 12:30:03] DEBUG[12438]: chan_sip.c:11610 handle_response_invite: SIP response 180 to standard invite [Feb 8 12:30:03] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4012-08a67c38 [Feb 8 12:30:03] -- SIP/4012-08a67c38 is ringing [Feb 8 12:30:03] <--- Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5d82e3176E478BFC;received=172.16.16.101 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Date: Thu, 08 Feb 2007 20:30:03 GMT Remote-Party-ID: "ITS Test 3" ;party=called;id-type=subscriber;screen=yes Content-Length: 0 <------------> [Feb 8 12:30:03] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:03] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:03] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4012 - state 6 (Ringing) [Feb 8 12:30:03] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:03] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:03] DEBUG[12505]: app_queue.c:546 changethread: Device 'SIP/4012' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Feb 8 12:30:05] <--- SIP read from 172.16.16.133:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK46db74e2;rport From: "ITS Test 2" ;tag=as054a4c33 To: ;tag=C207580-50CC03A7 CSeq: 102 INVITE Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1170966595 1170966595 IN IP4 172.16.16.133 s=Polycom IP Phone c=IN IP4 172.16.16.133 t=0 0 m=audio 2230 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK46db74e2;rport (63) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "ITS Test 2" ;tag=as054a4c33 (57) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=C207580-50CC03A7 (54) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 INVITE (16) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 (54) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Type: application/sdp (29) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Content-Length: 201 (19) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: (0) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: v=0 (3) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: o=- 1170966595 1170966595 IN IP4 172.16.16.133 (46) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: s=Polycom IP Phone (18) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: c=IN IP4 172.16.16.133 (22) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: t=0 0 (5) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: m=audio 2230 RTP/AVP 0 101 (26) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=sendrecv (10) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 8 12:30:05] --- (11 headers 9 lines) --- [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:2070 __sip_ack: Acked pending invite 102 [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '7b13f17e349ea65f031be7f1482226e2@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:11610 handle_response_invite: SIP response 200 to standard invite [Feb 8 12:30:05] Found RTP audio format 0 [Feb 8 12:30:05] Found RTP audio format 101 [Feb 8 12:30:05] Peer audio RTP is at port 172.16.16.133:2230 [Feb 8 12:30:05] Found description format PCMU for ID 0 [Feb 8 12:30:05] Found description format telephone-event for ID 101 [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:5105 process_sdp: T38 state changed to 0 on channel SIP/4012-08a67c38 [Feb 8 12:30:05] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 8 12:30:05] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 8 12:30:05] Peer audio RTP is at port 172.16.16.133:2230 [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:5185 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:5192 process_sdp: We have an owner, now see if we need to change this call [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:3002 update_call_counter: Updating call counter for outgoing call [Feb 8 12:30:05] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4012 [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:7953 build_route: build_route: Contact hop: [Feb 8 12:30:05] list_route: hop: [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:5622 reqprep: Strict routing enforced for session 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 [Feb 8 12:30:05] set_destination: Parsing for address/port to send to [Feb 8 12:30:05] set_destination: set destination to 172.16.16.133, port 5060 [Feb 8 12:30:05] Transmitting (no NAT) to 172.16.16.133:5060: ACK sip:4012@172.16.16.133:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK2af5b531;rport From: "ITS Test 2" ;tag=as054a4c33 To: ;tag=C207580-50CC03A7 Contact: Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:05] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:05] DEBUG[12501]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4012-08a67c38 [Feb 8 12:30:05] -- SIP/4012-08a67c38 answered SIP/4011-0897d348 [Feb 8 12:30:05] DEBUG[12501]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011-0897d348 [Feb 8 12:30:05] DEBUG[12501]: chan_sip.c:3455 sip_answer: SIP answering channel: SIP/4011-0897d348 [Feb 8 12:30:05] DEBUG[12501]: chan_sip.c:6412 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 8 12:30:05] DEBUG[12501]: chan_sip.c:6178 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Feb 8 12:30:05] DEBUG[12501]: chan_sip.c:6179 add_sdp: ** Our prefcodec: 0x0 (nothing) [Feb 8 12:30:05] Audio is at 172.16.16.12 port 19884 [Feb 8 12:30:05] Adding codec 0x4 (ulaw) to SDP [Feb 8 12:30:05] Adding non-codec 0x1 (telephone-event) to SDP [Feb 8 12:30:05] DEBUG[12501]: chan_sip.c:6310 add_sdp: -- Done with adding codecs to SDP [Feb 8 12:30:05] DEBUG[12501]: chan_sip.c:6355 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Feb 8 12:30:05] <--- Reliably Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5d82e3176E478BFC;received=172.16.16.101 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Remote-Party-ID: "ITS Test 3" ;party=called;id-type=subscriber;screen=yes Content-Type: application/sdp Content-Length: 240 v=0 o=root 12420 12420 IN IP4 172.16.16.12 s=session c=IN IP4 172.16.16.12 t=0 0 m=audio 19884 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Feb 8 12:30:05] DEBUG[12501]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #432 [Feb 8 12:30:05] -- Packet2Packet bridging SIP/4011-0897d348 and SIP/4012-08a67c38 [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4012 - state 2 (In use) [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:05] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:05] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:05] Reliably Transmitting (no NAT) to 172.16.16.135:5060: NOTIFY sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK51688627;rport From: ;tag=as7fcef0ca To: "4010" ;tag=B5A00269-C426A3B4 Contact: Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 367
--- [Feb 8 12:30:05] DEBUG[12424]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #433 [Feb 8 12:30:05] Extension Changed 4012 new state InUse for Notify User 4010 [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:05] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4012 - state 2 (In use) [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:05] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:05] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 2 (In use) [Feb 8 12:30:05] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:05] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:05] DEBUG[12506]: app_queue.c:546 changethread: Device 'SIP/4012' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:05] DEBUG[12507]: app_queue.c:546 changethread: Device 'SIP/4012' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:05] DEBUG[12508]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:05] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK51688627;rport From: ;tag=as7fcef0ca To: "4010" ;tag=B5A00269-C426A3B4 CSeq: 104 NOTIFY Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK51688627;rport (63) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=as7fcef0ca (53) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "4010" ;tag=B5A00269-C426A3B4 (65) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 104 NOTIFY (16) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 (49) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: presence (15) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:05] --- (10 headers 0 lines) --- [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #433 [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on 'ae75b12d-d9314ac3-30e8f7c6@172.16.16.135' of Request 104: Match Not Found [Feb 8 12:30:05] SIP Response message for INCOMING dialog NOTIFY arrived [Feb 8 12:30:05] <--- SIP read from 172.16.16.101:5060 ---> ACK sip:4012@172.16.16.12 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK64b20e908334E6F5 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c CSeq: 2 ACK Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="a34a604bd4f802b2aa2258171262e82c", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: ACK sip:4012@172.16.16.12 SIP/2.0 (33) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK64b20e908334E6F5 (66) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=225C83B2-C4872DCD (67) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=as0083934c (62) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 2 ACK (11) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="a34a604bd4f802b2aa2258171262e82c", algorithm=MD5 (188) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Max-Forwards: 70 (16) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 0 (17) [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:05] --- (12 headers 0 lines) --- [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #432 [Feb 8 12:30:05] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '5588e51e-c415a168-c97a9e83@172.16.16.101' of Response 2: Match Not Found [Feb 8 12:30:06] <--- SIP read from 172.16.16.101:5060 ---> INVITE sip:4012@172.16.16.12 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKa90588daD911343F From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c CSeq: 3 INVITE Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="27183e1cd8d1fc15dbd7213380579dd8", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1170966594 1170966595 IN IP4 172.16.16.101 s=Polycom IP Phone c=IN IP4 172.16.16.101 t=0 0 m=audio 2238 RTP/AVP 0 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: INVITE sip:4012@172.16.16.12 SIP/2.0 (36) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKa90588daD911343F (66) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=225C83B2-C4872DCD (67) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=as0083934c (62) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 3 INVITE (14) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Supported: 100rel,replaces (26) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow-Events: talk,hold,conference (34) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="27183e1cd8d1fc15dbd7213380579dd8", algorithm=MD5 (188) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Max-Forwards: 70 (16) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 13: Content-Type: application/sdp (29) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 14: Content-Length: 201 (19) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 15: (0) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: v=0 (3) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: o=- 1170966594 1170966595 IN IP4 172.16.16.101 (46) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: s=Polycom IP Phone (18) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: c=IN IP4 172.16.16.101 (22) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: t=0 0 (5) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: m=audio 2238 RTP/AVP 0 101 (26) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=sendonly (10) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 8 12:30:06] --- (15 headers 9 lines) --- [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 8 12:30:06] Sending to 172.16.16.101 : 5060 (no NAT) [Feb 8 12:30:06] Found RTP audio format 0 [Feb 8 12:30:06] Found RTP audio format 101 [Feb 8 12:30:06] Peer audio RTP is at port 172.16.16.101:2238 [Feb 8 12:30:06] Found description format PCMU for ID 0 [Feb 8 12:30:06] Found description format telephone-event for ID 101 [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:5105 process_sdp: T38 state changed to 0 on channel SIP/4011-0897d348 [Feb 8 12:30:06] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 8 12:30:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 8 12:30:06] Peer audio RTP is at port 172.16.16.101:2238 [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:5185 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:5192 process_sdp: We have an owner, now see if we need to change this call [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:13416 handle_request_invite: Got a SIP re-invite for call 5588e51e-c415a168-c97a9e83@172.16.16.101 [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:13511 handle_request_invite: SIP/4011-0897d348: This call is UP.... [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:6412 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:6178 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:6179 add_sdp: ** Our prefcodec: 0x0 (nothing) [Feb 8 12:30:06] Audio is at 172.16.16.12 port 19884 [Feb 8 12:30:06] Adding codec 0x4 (ulaw) to SDP [Feb 8 12:30:06] Adding non-codec 0x1 (telephone-event) to SDP [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:6310 add_sdp: -- Done with adding codecs to SDP [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:6355 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Feb 8 12:30:06] <--- Reliably Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKa90588daD911343F;received=172.16.16.101 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Remote-Party-ID: "ITS Test 3" ;party=called;id-type=subscriber;screen=yes Content-Type: application/sdp Content-Length: 240 v=0 o=root 12420 12421 IN IP4 172.16.16.12 s=session c=IN IP4 172.16.16.12 t=0 0 m=audio 19884 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #434 [Feb 8 12:30:06] DEBUG[12501]: channel.c:2838 set_format: Set channel SIP/4012-08a67c38 to write format slin [Feb 8 12:30:06] -- Started music on hold, class 'default', on channel 'SIP/4012-08a67c38' [Feb 8 12:30:06] DEBUG[12501]: channel.c:1991 ast_settimeout: Scheduling timer at 160 sample intervals [Feb 8 12:30:06] DEBUG[12501]: channel.c:2325 __ast_read: Generator got voice, switching to phase locked mode [Feb 8 12:30:06] DEBUG[12501]: channel.c:1991 ast_settimeout: Scheduling timer at 0 sample intervals [Feb 8 12:30:06] DEBUG[12501]: channel.c:2334 __ast_read: Auto-deactivating generator [Feb 8 12:30:06] DEBUG[12501]: channel.c:2838 set_format: Set channel SIP/4012-08a67c38 to write format ulaw [Feb 8 12:30:06] -- Stopped music on hold on SIP/4012-08a67c38 [Feb 8 12:30:06] DEBUG[12501]: channel.c:1991 ast_settimeout: Scheduling timer at 0 sample intervals [Feb 8 12:30:06] <--- SIP read from 172.16.16.101:5060 ---> ACK sip:4012@172.16.16.12 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKea529248D6EA789 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c CSeq: 3 ACK Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Max-Forwards: 70 Content-Length: 0 <-------------> [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: ACK sip:4012@172.16.16.12 SIP/2.0 (33) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKea529248D6EA789 (65) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=225C83B2-C4872DCD (67) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=as0083934c (62) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 3 ACK (11) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Content-Length: 0 (17) [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: (0) [Feb 8 12:30:06] --- (11 headers 0 lines) --- [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #434 [Feb 8 12:30:06] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '5588e51e-c415a168-c97a9e83@172.16.16.101' of Response 3: Match Not Found [Feb 8 12:30:08] <--- SIP read from 172.16.16.101:5060 ---> INVITE sip:4010@voip.dogmatix.dnv.org;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK317aef6eBF3740D3 From: "4011" ;tag=FA65264C-B4B1E567 To: CSeq: 1 INVITE Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1170966600 1170966600 IN IP4 172.16.16.101 s=Polycom IP Phone c=IN IP4 172.16.16.101 t=0 0 m=audio 2240 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: INVITE sip:4010@voip.dogmatix.dnv.org;user=phone SIP/2.0 (56) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK317aef6eBF3740D3 (66) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=FA65264C-B4B1E567 (67) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (47) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1 INVITE (14) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 (49) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Supported: 100rel,replaces (26) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow-Events: talk,hold,conference (34) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Max-Forwards: 70 (16) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 13: Content-Length: 239 (19) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 14: (0) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: v=0 (3) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: o=- 1170966600 1170966600 IN IP4 172.16.16.101 (46) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: s=Polycom IP Phone (18) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: c=IN IP4 172.16.16.101 (22) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: t=0 0 (5) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: m=audio 2240 RTP/AVP 0 8 18 101 (31) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 8 12:30:08] --- (14 headers 10 lines) --- [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:2574 do_setnat: Setting NAT on RTP to Off [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for 3370dbb8-bc70ee02-1355001d@172.16.16.101 - INVITE (With RTP) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:1679 parse_sip_options: Begin: parsing SIP "Supported: 100rel,replaces" [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:1687 parse_sip_options: Found SIP option: -100rel- [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:1693 parse_sip_options: Matched SIP option: 100rel [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:1687 parse_sip_options: Found SIP option: -replaces- [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:1693 parse_sip_options: Matched SIP option: replaces [Feb 8 12:30:08] Sending to 172.16.16.101 : 5060 (no NAT) [Feb 8 12:30:08] Using INVITE request as basis request - 3370dbb8-bc70ee02-1355001d@172.16.16.101 [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:2574 do_setnat: Setting NAT on RTP to Off [Feb 8 12:30:08] <--- Reliably Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK317aef6eBF3740D3;received=172.16.16.101 From: "4011" ;tag=FA65264C-B4B1E567 To: ;tag=as22a9436c Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="640bce0b" Content-Length: 0 <------------> [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #436 [Feb 8 12:30:08] Scheduling destruction of SIP dialog '3370dbb8-bc70ee02-1355001d@172.16.16.101' in 32000 ms (Method: INVITE) [Feb 8 12:30:08] Found user '4011' [Feb 8 12:30:08] <--- SIP read from 172.16.16.101:5060 ---> ACK sip:4010@voip.dogmatix.dnv.org SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK317aef6eBF3740D3 From: "4011" ;tag=FA65264C-B4B1E567 To: ;tag=as22a9436c CSeq: 1 ACK Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Max-Forwards: 70 Content-Length: 0 <-------------> [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: ACK sip:4010@voip.dogmatix.dnv.org SIP/2.0 (42) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK317aef6eBF3740D3 (66) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=FA65264C-B4B1E567 (67) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=as22a9436c (62) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1 ACK (11) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 (49) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Content-Length: 0 (17) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: (0) [Feb 8 12:30:08] --- (11 headers 0 lines) --- [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #436 [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '3370dbb8-bc70ee02-1355001d@172.16.16.101' of Response 1: Match Not Found [Feb 8 12:30:08] <--- SIP read from 172.16.16.101:5060 ---> INVITE sip:4010@voip.dogmatix.dnv.org;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5437f0b1A1378496 From: "4011" ;tag=FA65264C-B4B1E567 To: CSeq: 2 INVITE Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="640bce0b", uri="sip:4010@voip.dogmatix.dnv.org;user=phone", response="a1dec169a891efada75bf7e28ec04807", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1170966600 1170966600 IN IP4 172.16.16.101 s=Polycom IP Phone c=IN IP4 172.16.16.101 t=0 0 m=audio 2240 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: INVITE sip:4010@voip.dogmatix.dnv.org;user=phone SIP/2.0 (56) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5437f0b1A1378496 (66) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=FA65264C-B4B1E567 (67) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (47) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 2 INVITE (14) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 (49) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Supported: 100rel,replaces (26) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow-Events: talk,hold,conference (34) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="640bce0b", uri="sip:4010@voip.dogmatix.dnv.org;user=phone", response="a1dec169a891efada75bf7e28ec04807", algorithm=MD5 (188) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Max-Forwards: 70 (16) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 13: Content-Type: application/sdp (29) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 14: Content-Length: 239 (19) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 15: (0) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: v=0 (3) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: o=- 1170966600 1170966600 IN IP4 172.16.16.101 (46) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: s=Polycom IP Phone (18) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: c=IN IP4 172.16.16.101 (22) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: t=0 0 (5) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: m=audio 2240 RTP/AVP 0 8 18 101 (31) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:18 G729/8000 (21) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 8 12:30:08] --- (15 headers 10 lines) --- [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Feb 8 12:30:08] Sending to 172.16.16.101 : 5060 (no NAT) [Feb 8 12:30:08] Using INVITE request as basis request - 3370dbb8-bc70ee02-1355001d@172.16.16.101 [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:2574 do_setnat: Setting NAT on RTP to Off [Feb 8 12:30:08] Found user '4011' [Feb 8 12:30:08] Found RTP audio format 0 [Feb 8 12:30:08] Found RTP audio format 8 [Feb 8 12:30:08] Found RTP audio format 18 [Feb 8 12:30:08] Found RTP audio format 101 [Feb 8 12:30:08] Peer audio RTP is at port 172.16.16.101:2240 [Feb 8 12:30:08] Found description format PCMU for ID 0 [Feb 8 12:30:08] Found description format PCMA for ID 8 [Feb 8 12:30:08] Found description format G729 for ID 18 [Feb 8 12:30:08] Found description format telephone-event for ID 101 [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:5105 process_sdp: T38 state changed to 0 on channel [Feb 8 12:30:08] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 8 12:30:08] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 8 12:30:08] Peer audio RTP is at port 172.16.16.101:2240 [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:5185 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:13363 handle_request_invite: Checking SIP call limits for device 4011 [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:3002 update_call_counter: Updating call counter for incoming call [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:3076 update_call_counter: Call from peer '4011' is 2 out of 200 [Feb 8 12:30:08] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011 [Feb 8 12:30:08] Looking for 4010 in internal (domain voip.dogmatix.dnv.org) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:3802 sip_new: *** Our native formats are 0x4 (ulaw) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:3803 sip_new: *** Joint capabilities are 0x4 (ulaw) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:3804 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:3805 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:3828 sip_new: This channel will not be able to handle video. [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:7953 build_route: build_route: Contact hop: [Feb 8 12:30:08] list_route: hop: [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:13438 handle_request_invite: SIP/4011-08a6bf60: New call is still down.... Trying... [Feb 8 12:30:08] <--- Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5437f0b1A1378496;received=172.16.16.101 From: "4011" ;tag=FA65264C-B4B1E567 To: Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:08] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011-08a6bf60 [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:08] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 2 (In use) [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:08] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:08] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 2 (In use) [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:08] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'Goto' [Feb 8 12:30:08] -- Executing [4010@internal:1] Goto("SIP/4011-08a6bf60", "ITS|4010|1") in new stack [Feb 8 12:30:08] -- Goto (ITS,4010,1) [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '4011' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'Macro' [Feb 8 12:30:08] -- Executing [4010@ITS:1] Macro("SIP/4011-08a6bf60", "dnv-incoming-call-handler|4011|4010") in new stack [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is 'ITS Test 1' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'NoOp' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:1] NoOp("SIP/4011-08a6bf60", "ITS Test 1") in new stack [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is 'ITS Test 1' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:2] GotoIf("SIP/4011-08a6bf60", "0?SetVoiceMailContext") in new stack [Feb 8 12:30:08] DEBUG[12510]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:08] DEBUG[12511]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:08] DEBUG[12509]: pbx.c:5947 pbx_builtin_gotoif: Not taking any branch [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is 'SIP' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '1' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:3] GotoIf("SIP/4011-08a6bf60", "1?:SetVoiceMailContext") in new stack [Feb 8 12:30:08] DEBUG[12509]: pbx.c:5947 pbx_builtin_gotoif: Not taking any branch [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is 'ITS Test 1' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'SIPCalledRPID' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:4] SIPCalledRPID("SIP/4011-08a6bf60", "ITS Test 1|4010") in new stack [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:5] GotoIf("SIP/4011-08a6bf60", "0?:NVDPL") in new stack [Feb 8 12:30:08] -- Goto (macro-dnv-incoming-call-handler,s,8) [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:8] GotoIf("SIP/4011-08a6bf60", "0?:Default") in new stack [Feb 8 12:30:08] -- Goto (macro-dnv-incoming-call-handler,s,12) [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'Set' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:12] Set("SIP/4011-08a6bf60", "vm_context=default") in new stack [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '1' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '1' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:13] GotoIf("SIP/4011-08a6bf60", "0?:Proceed") in new stack [Feb 8 12:30:08] -- Goto (macro-dnv-incoming-call-handler,s,19) [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '"ITS Test 2" <4011>' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'NoOp' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:19] NoOp("SIP/4011-08a6bf60", ""ITS Test 2" <4011>") in new stack [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '4011' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:20] GotoIf("SIP/4011-08a6bf60", "0?:IdentifyCID") in new stack [Feb 8 12:30:08] -- Goto (macro-dnv-incoming-call-handler,s,24) [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '4011' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1690 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'GotoIf' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:24] GotoIf("SIP/4011-08a6bf60", "0?:Finish") in new stack [Feb 8 12:30:08] -- Goto (macro-dnv-incoming-call-handler,s,26) [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1622 pbx_substitute_variables_helper_full: Function result is '"ITS Test 2" <4011>' [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'NoOp' [Feb 8 12:30:08] -- Executing [s@macro-dnv-incoming-call-handler:26] NoOp("SIP/4011-08a6bf60", ""ITS Test 2" <4011>") in new stack [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'Macro' [Feb 8 12:30:08] -- Executing [4010@ITS:2] Macro("SIP/4011-08a6bf60", "dnv-standard-dialer|SIP/4010|20|o") in new stack [Feb 8 12:30:08] DEBUG[12509]: pbx.c:1769 pbx_extension_helper: Launching 'Dial' [Feb 8 12:30:08] -- Executing [s@macro-dnv-standard-dialer:1] Dial("SIP/4011-08a6bf60", "SIP/4010|20|o") in new stack [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:15246 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:2574 do_setnat: Setting NAT on RTP to Off [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:3802 sip_new: *** Our native formats are 0x4 (ulaw) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:3803 sip_new: *** Joint capabilities are 0x0 (nothing) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:3804 sip_new: *** Our capabilities are 0x6 (gsm|ulaw) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:3805 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:3807 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:3828 sip_new: This channel will not be able to handle video. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-standard-dialer-s-1. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable ARG3. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable ARG2. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable ARG1. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable MACRO_PRIORITY. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable MACRO_CONTEXT. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable MACRO_EXTEN. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-ITS-4010-2. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-26. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-24. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-20. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-19. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-13. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable vm_context. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-12. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-8. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-5. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-4. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-3. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-2. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-macro-dnv-incoming-call-handler-s-1. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-ITS-4010-1. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable STACK-internal-4010-1. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Feb 8 12:30:08] DEBUG[12509]: channel.c:3294 ast_channel_inherit_variables: Not copying variable SIPURI. [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:2829 sip_call: Outgoing Call for 4010 [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:3002 update_call_counter: Updating call counter for outgoing call [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:3076 update_call_counter: Call to peer '4010' is 1 out of 200 [Feb 8 12:30:08] DEBUG[12509]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4010 [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:08] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4010 - state 6 (Ringing) [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:08] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:08] DEBUG[12512]: app_queue.c:546 changethread: Device 'SIP/4010' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:2844 sip_call: Our T38 capability (0), joint T38 capability (0) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:6178 add_sdp: ** Our capability: 0x6 (gsm|ulaw) Video flag: False [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:6179 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Feb 8 12:30:08] Audio is at 172.16.16.12 port 12222 [Feb 8 12:30:08] Adding codec 0x4 (ulaw) to SDP [Feb 8 12:30:08] Adding codec 0x2 (gsm) to SDP [Feb 8 12:30:08] Adding non-codec 0x1 (telephone-event) to SDP [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:6310 add_sdp: -- Done with adding codecs to SDP [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:6355 add_sdp: Done building SDP. Settling with this capability: 0x6 (gsm|ulaw) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 0: INVITE sip:4010@172.16.16.135:5060 SIP/2.0 (42) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1d8c9787;rport (63) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 2: From: "ITS Test 2" ;tag=as3d35352a (57) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 4: Contact: (32) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 (54) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 INVITE (16) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:08 GMT (35) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 12: Content-Type: application/sdp (29) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 13: Content-Length: 263 (19) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4562 parse_request: Header 14: (0) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: v=0 (3) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: o=root 12420 12420 IN IP4 172.16.16.12 (38) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: s=session (9) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: c=IN IP4 172.16.16.12 (21) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: t=0 0 (5) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: m=audio 12222 RTP/AVP 0 3 101 (29) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: a=rtpmap:3 GSM/8000 (19) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: a=fmtp:101 0-16 (15) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: a=silenceSupp:off - - - - (25) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: a=ptime:20 (10) [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:4594 parse_request: Line: a=sendrecv (10) [Feb 8 12:30:08] Reliably Transmitting (no NAT) to 172.16.16.135:5060: INVITE sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1d8c9787;rport From: "ITS Test 2" ;tag=as3d35352a To: Contact: Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=root 12420 12420 IN IP4 172.16.16.12 s=session c=IN IP4 172.16.16.12 t=0 0 m=audio 12222 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Feb 8 12:30:08] DEBUG[12509]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #438 [Feb 8 12:30:08] -- Called 4010 [Feb 8 12:30:08] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1d8c9787;rport From: "ITS Test 2" ;tag=as3d35352a To: ;tag=52688A3-9204426 CSeq: 102 INVITE Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 100 Trying (18) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1d8c9787;rport (63) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "ITS Test 2" ;tag=as3d35352a (57) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=52688A3-9204426 (53) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 INVITE (16) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 (54) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Content-Length: 0 (17) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: (0) [Feb 8 12:30:08] --- (9 headers 0 lines) --- [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:2121 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #438 - INVITE (got response) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:2130 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12' Request 102: Found [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:11610 handle_response_invite: SIP response 100 to standard invite [Feb 8 12:30:08] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1d8c9787;rport From: "ITS Test 2" ;tag=as3d35352a To: ;tag=52688A3-9204426 CSeq: 102 INVITE Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1d8c9787;rport (63) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "ITS Test 2" ;tag=as3d35352a (57) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=52688A3-9204426 (53) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 INVITE (16) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 (54) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Allow-Events: talk,hold,conference (34) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:08] --- (10 headers 0 lines) --- [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:2130 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12' Request 102: Found [Feb 8 12:30:08] DEBUG[12438]: chan_sip.c:11610 handle_response_invite: SIP response 180 to standard invite [Feb 8 12:30:08] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4010-08a706c8 [Feb 8 12:30:08] -- SIP/4010-08a706c8 is ringing [Feb 8 12:30:08] <--- Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5437f0b1A1378496;received=172.16.16.101 From: "4011" ;tag=FA65264C-B4B1E567 To: ;tag=as46fc0c03 Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Date: Thu, 08 Feb 2007 20:30:08 GMT Remote-Party-ID: "ITS Test 1" ;party=called;id-type=subscriber;screen=yes Content-Length: 0 <------------> [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:08] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4010 - state 6 (Ringing) [Feb 8 12:30:08] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:08] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:08] DEBUG[12513]: app_queue.c:546 changethread: Device 'SIP/4010' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. [Feb 8 12:30:09] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1d8c9787;rport From: "ITS Test 2" ;tag=as3d35352a To: ;tag=52688A3-9204426 CSeq: 102 INVITE Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1170966573 1170966573 IN IP4 172.16.16.135 s=Polycom IP Phone c=IN IP4 172.16.16.135 t=0 0 m=audio 2250 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1d8c9787;rport (63) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "ITS Test 2" ;tag=as3d35352a (57) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=52688A3-9204426 (53) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 INVITE (16) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 (54) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Type: application/sdp (29) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Content-Length: 201 (19) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: (0) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: v=0 (3) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: o=- 1170966573 1170966573 IN IP4 172.16.16.135 (46) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: s=Polycom IP Phone (18) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: c=IN IP4 172.16.16.135 (22) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: t=0 0 (5) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: m=audio 2250 RTP/AVP 0 101 (26) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=sendrecv (10) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Feb 8 12:30:09] --- (11 headers 9 lines) --- [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:2070 __sip_ack: Acked pending invite 102 [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:11610 handle_response_invite: SIP response 200 to standard invite [Feb 8 12:30:09] Found RTP audio format 0 [Feb 8 12:30:09] Found RTP audio format 101 [Feb 8 12:30:09] Peer audio RTP is at port 172.16.16.135:2250 [Feb 8 12:30:09] Found description format PCMU for ID 0 [Feb 8 12:30:09] Found description format telephone-event for ID 101 [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:5105 process_sdp: T38 state changed to 0 on channel SIP/4010-08a706c8 [Feb 8 12:30:09] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) [Feb 8 12:30:09] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Feb 8 12:30:09] Peer audio RTP is at port 172.16.16.135:2250 [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:5185 process_sdp: We're settling with these formats: 0x4 (ulaw) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:5192 process_sdp: We have an owner, now see if we need to change this call [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:3002 update_call_counter: Updating call counter for outgoing call [Feb 8 12:30:09] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4010 [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:7953 build_route: build_route: Contact hop: [Feb 8 12:30:09] list_route: hop: [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:5622 reqprep: Strict routing enforced for session 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 [Feb 8 12:30:09] set_destination: Parsing for address/port to send to [Feb 8 12:30:09] set_destination: set destination to 172.16.16.135, port 5060 [Feb 8 12:30:09] Transmitting (no NAT) to 172.16.16.135:5060: ACK sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK12f2d320;rport From: "ITS Test 2" ;tag=as3d35352a To: ;tag=52688A3-9204426 Contact: Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 8 12:30:09] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:09] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:09] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4010 - state 2 (In use) [Feb 8 12:30:09] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:09] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:09] DEBUG[12509]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4010-08a706c8 [Feb 8 12:30:09] -- SIP/4010-08a706c8 answered SIP/4011-08a6bf60 [Feb 8 12:30:09] DEBUG[12509]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011-08a6bf60 [Feb 8 12:30:09] DEBUG[12509]: chan_sip.c:3455 sip_answer: SIP answering channel: SIP/4011-08a6bf60 [Feb 8 12:30:09] DEBUG[12509]: chan_sip.c:6412 transmit_response_with_sdp: Setting framing from config on incoming call [Feb 8 12:30:09] DEBUG[12509]: chan_sip.c:6178 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [Feb 8 12:30:09] DEBUG[12509]: chan_sip.c:6179 add_sdp: ** Our prefcodec: 0x0 (nothing) [Feb 8 12:30:09] Audio is at 172.16.16.12 port 10024 [Feb 8 12:30:09] Adding codec 0x4 (ulaw) to SDP [Feb 8 12:30:09] Adding non-codec 0x1 (telephone-event) to SDP [Feb 8 12:30:09] DEBUG[12509]: chan_sip.c:6310 add_sdp: -- Done with adding codecs to SDP [Feb 8 12:30:09] DEBUG[12509]: chan_sip.c:6355 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [Feb 8 12:30:09] <--- Reliably Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK5437f0b1A1378496;received=172.16.16.101 From: "4011" ;tag=FA65264C-B4B1E567 To: ;tag=as46fc0c03 Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Remote-Party-ID: "ITS Test 1" ;party=called;id-type=subscriber;screen=yes Content-Type: application/sdp Content-Length: 240 v=0 o=root 12420 12420 IN IP4 172.16.16.12 s=session c=IN IP4 172.16.16.12 t=0 0 m=audio 10024 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Feb 8 12:30:09] DEBUG[12509]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #440 [Feb 8 12:30:09] -- Packet2Packet bridging SIP/4011-08a6bf60 and SIP/4010-08a706c8 [Feb 8 12:30:09] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:09] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:09] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4010 - state 2 (In use) [Feb 8 12:30:09] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:09] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:09] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:09] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:09] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 2 (In use) [Feb 8 12:30:09] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:09] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:09] DEBUG[12514]: app_queue.c:546 changethread: Device 'SIP/4010' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:09] DEBUG[12515]: app_queue.c:546 changethread: Device 'SIP/4010' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:09] DEBUG[12516]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:09] <--- SIP read from 172.16.16.101:5060 ---> ACK sip:4010@172.16.16.12 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK971eb32aEAE4F68F From: "4011" ;tag=FA65264C-B4B1E567 To: ;tag=as46fc0c03 CSeq: 2 ACK Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="640bce0b", uri="sip:4010@voip.dogmatix.dnv.org;user=phone", response="a1dec169a891efada75bf7e28ec04807", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: ACK sip:4010@172.16.16.12 SIP/2.0 (33) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bK971eb32aEAE4F68F (66) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=FA65264C-B4B1E567 (67) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=as46fc0c03 (62) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 2 ACK (11) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 (49) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="640bce0b", uri="sip:4010@voip.dogmatix.dnv.org;user=phone", response="a1dec169a891efada75bf7e28ec04807", algorithm=MD5 (188) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Max-Forwards: 70 (16) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 0 (17) [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:09] --- (12 headers 0 lines) --- [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received ACK (6) - Command in SIP ACK [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #440 [Feb 8 12:30:09] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '3370dbb8-bc70ee02-1355001d@172.16.16.101' of Response 2: Match Not Found [Feb 8 12:30:11] DEBUG[12430]: chan_iax2.c:7099 socket_process: Peer gabriola-out: got pong, lastms 16, historicms 16, maxms 2000 [Feb 8 12:30:12] <--- SIP read from 172.16.16.101:5060 ---> REFER sip:4012@172.16.16.12 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKec078374E43399D9 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c CSeq: 4 REFER Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Refer-To: Referred-By: Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="a34a604bd4f802b2aa2258171262e82c", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: REFER sip:4012@172.16.16.12 SIP/2.0 (35) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKec078374E43399D9 (66) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=225C83B2-C4872DCD (67) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=as0083934c (62) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 4 REFER (13) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Refer-To: (158) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Referred-By: (45) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="a34a604bd4f802b2aa2258171262e82c", algorithm=MD5 (188) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Max-Forwards: 70 (16) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 13: (0) [Feb 8 12:30:12] --- (13 headers 0 lines) --- [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received REFER (9) - Command in SIP REFER [Feb 8 12:30:12] Call 5588e51e-c415a168-c97a9e83@172.16.16.101 got a SIP call transfer from caller: (REFER)! [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:8685 get_refer_info: Attended transfer: Will use Replace-Call-ID : 3370dbb8-bc70ee02-1355001d@172.16.16.101 (No check of from/to tags) [Feb 8 12:30:12] SIP transfer to extension 4010@ITS by 4011@voip.dogmatix.dnv.org [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:13916 handle_request_refer: SIP attended transfer: Transferer channel SIP/4011-0897d348, transferee channel SIP/4012-08a67c38 [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:13932 handle_request_refer: Got SIP transfer, applying to bridged peer 'SIP/4012-08a67c38' [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:8539 get_sip_pvt_byid_locked: Looking for callid 3370dbb8-bc70ee02-1355001d@172.16.16.101 (fromtag FA65264C-B4B1E567 totag as46fc0c03) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:8563 get_sip_pvt_byid_locked: Matched INCOMING call - their tag is FA65264C-B4B1E567 Our tag is as46fc0c03 [Feb 8 12:30:12] <--- Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKec078374E43399D9;received=172.16.16.101 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:13687 local_attended_transfer: SIP attended transfer: trying to bridge SIP/4011-08a6bf60 and SIP/4012-08a67c38 [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:12697 attempt_transfer: Sip transfer:-------------------- [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:12699 attempt_transfer: -- Transferer to PBX channel: SIP/4011-0897d348 State Up [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:12703 attempt_transfer: -- Transferer to PBX second channel (target): SIP/4011-08a6bf60 State Up [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:12707 attempt_transfer: -- Bridged call to transferee: SIP/4012-08a67c38 State Up [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:12711 attempt_transfer: -- Bridged call to transfer target: SIP/4010-08a706c8 State Up [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:12714 attempt_transfer: -- END Sip transfer:-------------------- [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:12722 attempt_transfer: SIP transfer: Four channels to handle [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:12753 attempt_transfer: SIP transfer: trying to masquerade SIP/4012-08a67c38 into SIP/4011-08a6bf60 [Feb 8 12:30:12] DEBUG[12438]: channel.c:3227 ast_channel_masquerade: Planning to masquerade channel SIP/4012-08a67c38 into the structure of SIP/4011-08a6bf60 [Feb 8 12:30:12] DEBUG[12438]: channel.c:3241 ast_channel_masquerade: Done planning to masquerade channel SIP/4012-08a67c38 into the structure of SIP/4011-08a6bf60 [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:12758 attempt_transfer: SIP transfer: Succeeded to masquerade channels. [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:5622 reqprep: Strict routing enforced for session 5588e51e-c415a168-c97a9e83@172.16.16.101 [Feb 8 12:30:12] set_destination: Parsing for address/port to send to [Feb 8 12:30:12] set_destination: set destination to 172.16.16.101, port 5060 [Feb 8 12:30:12] Reliably Transmitting (no NAT) to 172.16.16.101:5060: NOTIFY sip:4011@172.16.16.101:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK02dcdf66;rport From: ;tag=as0083934c To: "4011" ;tag=225C83B2-C4872DCD Contact: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=4 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #441 [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:13715 local_attended_transfer: SIP attended transfer: Unlocking channel SIP/4011-08a6bf60 [Feb 8 12:30:12] DEBUG[12509]: channel.c:3358 ast_do_masquerade: Actually Masquerading SIP/4012-08a67c38(6) into the structure of SIP/4011-08a6bf60(6) [Feb 8 12:30:12] DEBUG[12509]: channel.c:3370 ast_do_masquerade: Got clone lock for masquerade on 'SIP/4012-08a67c38' at 0x8983d70 [Feb 8 12:30:12] DEBUG[12509]: chan_sip.c:3580 sip_fixup: SIP Fixup: New owner for dialogue 3370dbb8-bc70ee02-1355001d@172.16.16.101: SIP/4012-08a67c38 (Old parent: SIP/4012-08a67c38) [Feb 8 12:30:12] DEBUG[12509]: chan_sip.c:3306 sip_hangup: Hangup call SIP/4012-08a67c38, SIP callid 3370dbb8-bc70ee02-1355001d@172.16.16.101) [Feb 8 12:30:12] DEBUG[12509]: chan_sip.c:3314 sip_hangup: update_call_counter(4011) - decrement call limit counter on hangup [Feb 8 12:30:12] DEBUG[12509]: chan_sip.c:3002 update_call_counter: Updating call counter for incoming call [Feb 8 12:30:12] DEBUG[12509]: chan_sip.c:3050 update_call_counter: Call from peer '4011' removed from call limit 200 [Feb 8 12:30:12] DEBUG[12509]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011 [Feb 8 12:30:12] Scheduling destruction of SIP dialog '3370dbb8-bc70ee02-1355001d@172.16.16.101' in 32000 ms (Method: ACK) [Feb 8 12:30:12] DEBUG[12509]: chan_sip.c:5622 reqprep: Strict routing enforced for session 3370dbb8-bc70ee02-1355001d@172.16.16.101 [Feb 8 12:30:12] set_destination: Parsing for address/port to send to [Feb 8 12:30:12] set_destination: set destination to 172.16.16.101, port 5060 [Feb 8 12:30:12] Reliably Transmitting (no NAT) to 172.16.16.101:5060: BYE sip:4011@172.16.16.101:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK15fa54b4;rport From: ;tag=as46fc0c03 To: "4011" ;tag=FA65264C-B4B1E567 Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 8 12:30:12] DEBUG[12509]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #443 [Feb 8 12:30:12] DEBUG[12509]: channel.c:3565 ast_do_masquerade: Putting channel SIP/4012-08a67c38 in 4/4 formats [Feb 8 12:30:12] DEBUG[12509]: chan_sip.c:3580 sip_fixup: SIP Fixup: New owner for dialogue 7b13f17e349ea65f031be7f1482226e2@172.16.16.12: SIP/4012-08a67c38 (Old parent: SIP/4011-08a6bf60) [Feb 8 12:30:12] DEBUG[12509]: channel.c:3601 ast_do_masquerade: Released clone lock on 'SIP/4011-08a6bf60' [Feb 8 12:30:12] DEBUG[12509]: channel.c:3611 ast_do_masquerade: Done Masquerading SIP/4012-08a67c38 (6) [Feb 8 12:30:12] DEBUG[12509]: rtp.c:3038 bridge_p2p_loop: Oooh, something is weird, backing out [Feb 8 12:30:12] -- Packet2Packet bridging SIP/4012-08a67c38 and SIP/4010-08a706c8 [Feb 8 12:30:12] DEBUG[12501]: rtp.c:3038 bridge_p2p_loop: Oooh, something is weird, backing out [Feb 8 12:30:12] DEBUG[12501]: channel.c:4012 ast_channel_bridge: Bridge stops because we're zombie or need a soft hangup: c0=SIP/4011-0897d348, c1=SIP/4011-08a6bf60, flags: No,No,Yes,Yes [Feb 8 12:30:12] DEBUG[12501]: channel.c:4111 ast_channel_bridge: Bridge stops bridging channels SIP/4011-0897d348 and SIP/4011-08a6bf60 [Feb 8 12:30:12] DEBUG[12501]: channel.c:1692 ast_hangup: Hanging up zombie 'SIP/4011-08a6bf60' [Feb 8 12:30:12] DEBUG[12501]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011-08a6bf60 [Feb 8 12:30:12] DEBUG[12501]: rtp.c:1474 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Feb 8 12:30:12] DEBUG[12501]: app_dial.c:1648 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Feb 8 12:30:12] DEBUG[12501]: app_macro.c:261 _macro_exec: Spawn extension (macro-dnv-standard-dialer,s,1) exited non-zero on 'SIP/4011-0897d348' in macro 'dnv-standard-dialer' [Feb 8 12:30:12] DEBUG[12501]: pbx.c:2367 __ast_pbx_run: Spawn extension (macro-dnv-standard-dialer,s,1) exited non-zero on 'SIP/4011-0897d348' [Feb 8 12:30:12] == Spawn extension (macro-dnv-standard-dialer, s, 1) exited non-zero on 'SIP/4011-0897d348' [Feb 8 12:30:12] DEBUG[12501]: channel.c:1687 ast_hangup: Hanging up channel 'SIP/4011-0897d348' [Feb 8 12:30:12] DEBUG[12501]: chan_sip.c:3291 sip_hangup: SIP Transfer: Not hanging up right now... Rescheduling hangup for 5588e51e-c415a168-c97a9e83@172.16.16.101. [Feb 8 12:30:12] Scheduling destruction of SIP dialog '5588e51e-c415a168-c97a9e83@172.16.16.101' in 32000 ms (Method: REFER) [Feb 8 12:30:12] DEBUG[12501]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011-0897d348 [Feb 8 12:30:12] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:12] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:12] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 2 (In use) [Feb 8 12:30:12] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:12] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:12] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:12] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:12] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 2 (In use) [Feb 8 12:30:12] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:12] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:12] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:12] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:12] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 2 (In use) [Feb 8 12:30:12] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:12] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:12] DEBUG[12517]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:12] DEBUG[12518]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:12] DEBUG[12519]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 8 12:30:12] DEBUG[12509]: rtp.c:2670 ast_rtp_write: Ooh, format changed from unknown to ulaw [Feb 8 12:30:12] DEBUG[12509]: rtp.c:2687 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Feb 8 12:30:12] <--- SIP read from 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK02dcdf66;rport From: ;tag=as0083934c To: "4011" ;tag=225C83B2-C4872DCD CSeq: 102 NOTIFY Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 Contact: Event: refer;id=4 User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK02dcdf66;rport (63) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=as0083934c (64) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "4011" ;tag=225C83B2-C4872DCD (65) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 NOTIFY (16) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: refer;id=4 (17) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:12] --- (10 headers 0 lines) --- [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #441 [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '5588e51e-c415a168-c97a9e83@172.16.16.101' of Request 102: Match Not Found [Feb 8 12:30:12] SIP Response message for INCOMING dialog NOTIFY arrived [Feb 8 12:30:12] <--- SIP read from 172.16.16.101:5060 ---> BYE sip:4012@172.16.16.12 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKb57679be2A3D6323 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c CSeq: 5 BYE Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="9ae32455f3720061015d045e7d568455", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: BYE sip:4012@172.16.16.12 SIP/2.0 (33) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKb57679be2A3D6323 (66) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "4011" ;tag=225C83B2-C4872DCD (67) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=as0083934c (62) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 5 BYE (11) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Proxy-Authorization: Digest username="4011", realm="asterisk", nonce="1269c090", uri="sip:4012@voip.dogmatix.dnv.org;user=phone", response="9ae32455f3720061015d045e7d568455", algorithm=MD5 (188) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Content-Length: 0 (17) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: (0) [Feb 8 12:30:12] --- (11 headers 0 lines) --- [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received BYE (8) - Command in SIP BYE [Feb 8 12:30:12] Sending to 172.16.16.101 : 5060 (no NAT) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:1632 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 5588e51e-c415a168-c97a9e83@172.16.16.101 [Feb 8 12:30:12] Scheduling destruction of SIP dialog '5588e51e-c415a168-c97a9e83@172.16.16.101' in 32000 ms (Method: BYE) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:14156 handle_request_bye: Received bye, no owner, selfdestruct soon. [Feb 8 12:30:12] <--- Transmitting (no NAT) to 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.101:5060;branch=z9hG4bKb57679be2A3D6323;received=172.16.16.101 From: "4011" ;tag=225C83B2-C4872DCD To: ;tag=as0083934c Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 CSeq: 5 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:12] <--- SIP read from 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK15fa54b4;rport From: ;tag=as46fc0c03 To: "4011" ;tag=FA65264C-B4B1E567 CSeq: 102 BYE Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK15fa54b4;rport (63) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=as46fc0c03 (64) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "4011" ;tag=FA65264C-B4B1E567 (65) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 BYE (13) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 3370dbb8-bc70ee02-1355001d@172.16.16.101 (49) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Content-Length: 0 (17) [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: (0) [Feb 8 12:30:12] --- (9 headers 0 lines) --- [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #443 [Feb 8 12:30:12] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '3370dbb8-bc70ee02-1355001d@172.16.16.101' of Request 102: Match Not Found [Feb 8 12:30:12] SIP Response message for INCOMING dialog BYE arrived [Feb 8 12:30:12] Really destroying SIP dialog '3370dbb8-bc70ee02-1355001d@172.16.16.101' Method: ACK [Feb 8 12:30:15] DEBUG[12509]: rtp.c:871 ast_rtcp_read: Got RTCP report of 84 bytes [Feb 8 12:30:17] <--- SIP read from 172.16.16.133:5060 ---> BYE sip:4011@172.16.16.12 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.133:5060;branch=z9hG4bKd2a92f05E8891E46 From: ;tag=C207580-50CC03A7 To: "ITS Test 2" ;tag=as054a4c33 CSeq: 1 BYE Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Max-Forwards: 70 Content-Length: 0 <-------------> [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: BYE sip:4011@172.16.16.12 SIP/2.0 (33) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.133:5060;branch=z9hG4bKd2a92f05E8891E46 (66) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=C207580-50CC03A7 (56) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "ITS Test 2" ;tag=as054a4c33 (55) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1 BYE (11) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 (54) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:17] --- (10 headers 0 lines) --- [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received BYE (8) - Command in SIP BYE [Feb 8 12:30:17] Sending to 172.16.16.133 : 5060 (no NAT) [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:1632 sip_alreadygone: Setting SIP_ALREADYGONE on dialog 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 [Feb 8 12:30:17] DEBUG[12438]: chan_sip.c:14152 handle_request_bye: Received bye, issuing owner hangup [Feb 8 12:30:17] <--- Transmitting (no NAT) to 172.16.16.133:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.133:5060;branch=z9hG4bKd2a92f05E8891E46;received=172.16.16.133 From: ;tag=C207580-50CC03A7 To: "ITS Test 2" ;tag=as054a4c33 Call-ID: 7b13f17e349ea65f031be7f1482226e2@172.16.16.12 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:17] DEBUG[12509]: rtp.c:3069 bridge_p2p_loop: Oooh, got a hangup [Feb 8 12:30:17] DEBUG[12509]: channel.c:4041 ast_channel_bridge: Returning from native bridge, channels: SIP/4012-08a67c38, SIP/4010-08a706c8 [Feb 8 12:30:17] DEBUG[12509]: channel.c:1687 ast_hangup: Hanging up channel 'SIP/4010-08a706c8' [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:3306 sip_hangup: Hangup call SIP/4010-08a706c8, SIP callid 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12) [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:3314 sip_hangup: update_call_counter(4010) - decrement call limit counter on hangup [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:3002 update_call_counter: Updating call counter for outgoing call [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:3050 update_call_counter: Call to peer '4010' removed from call limit 200 [Feb 8 12:30:17] DEBUG[12509]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4010 [Feb 8 12:30:17] Scheduling destruction of SIP dialog '20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12' in 6400 ms (Method: INVITE) [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:5622 reqprep: Strict routing enforced for session 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 [Feb 8 12:30:17] set_destination: Parsing for address/port to send to [Feb 8 12:30:17] set_destination: set destination to 172.16.16.135, port 5060 [Feb 8 12:30:17] Reliably Transmitting (no NAT) to 172.16.16.135:5060: BYE sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1499ff8c;rport From: "ITS Test 2" ;tag=as3d35352a To: ;tag=52688A3-9204426 Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #448 [Feb 8 12:30:17] DEBUG[12509]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4010-08a706c8 [Feb 8 12:30:17] DEBUG[12509]: rtp.c:1474 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Feb 8 12:30:17] DEBUG[12509]: app_dial.c:1648 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Feb 8 12:30:17] DEBUG[12509]: app_macro.c:261 _macro_exec: Spawn extension (macro-dnv-standard-dialer,s,1) exited non-zero on 'SIP/4012-08a67c38' in macro 'dnv-standard-dialer' [Feb 8 12:30:17] DEBUG[12509]: pbx.c:2367 __ast_pbx_run: Spawn extension (macro-dnv-standard-dialer,s,1) exited non-zero on 'SIP/4012-08a67c38' [Feb 8 12:30:17] == Spawn extension (macro-dnv-standard-dialer, s, 1) exited non-zero on 'SIP/4012-08a67c38' [Feb 8 12:30:17] DEBUG[12509]: channel.c:1687 ast_hangup: Hanging up channel 'SIP/4012-08a67c38' [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:3306 sip_hangup: Hangup call SIP/4012-08a67c38, SIP callid 7b13f17e349ea65f031be7f1482226e2@172.16.16.12) [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:3314 sip_hangup: update_call_counter(4012) - decrement call limit counter on hangup [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:3002 update_call_counter: Updating call counter for outgoing call [Feb 8 12:30:17] DEBUG[12509]: chan_sip.c:3050 update_call_counter: Call to peer '4012' removed from call limit 200 [Feb 8 12:30:17] DEBUG[12509]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4012 [Feb 8 12:30:17] DEBUG[12509]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4012-08a67c38 [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4010 - state 1 (Not in use) [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4010 - state 1 (Not in use) [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4010 [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4010 [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4012 - state 1 (Not in use) [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:17] Reliably Transmitting (no NAT) to 172.16.16.135:5060: NOTIFY sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK74dc7e63;rport From: ;tag=as7fcef0ca To: "4010" ;tag=B5A00269-C426A3B4 Contact: Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 362
--- [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #449 [Feb 8 12:30:17] Extension Changed 4012 new state Idle for Notify User 4010 [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4012 - state 1 (Not in use) [Feb 8 12:30:17] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4012 [Feb 8 12:30:17] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4012 [Feb 8 12:30:17] DEBUG[12523]: app_queue.c:546 changethread: Device 'SIP/4010' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 8 12:30:17] DEBUG[12524]: app_queue.c:546 changethread: Device 'SIP/4010' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 8 12:30:17] DEBUG[12525]: app_queue.c:546 changethread: Device 'SIP/4012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 8 12:30:17] DEBUG[12526]: app_queue.c:546 changethread: Device 'SIP/4012' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 8 12:30:18] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1499ff8c;rport From: "ITS Test 2" ;tag=as3d35352a To: ;tag=52688A3-9204426 CSeq: 103 BYE Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK1499ff8c;rport (63) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "ITS Test 2" ;tag=as3d35352a (57) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=52688A3-9204426 (53) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 103 BYE (13) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12 (54) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Content-Length: 0 (17) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: (0) [Feb 8 12:30:18] --- (9 headers 0 lines) --- [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #448 [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12' of Request 103: Match Not Found [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:1865 retrans_pkt: SIP TIMER: Rescheduling retransmission #449 (1) NOTIFY - 4 [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:1879 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 78 ms (t1 39 ms (Retrans id #449)) [Feb 8 12:30:18] Retransmitting #1 (no NAT) to 172.16.16.135:5060: NOTIFY sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK74dc7e63;rport From: ;tag=as7fcef0ca To: "4010" ;tag=B5A00269-C426A3B4 Contact: Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 362
--- [Feb 8 12:30:18] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK74dc7e63;rport From: ;tag=as7fcef0ca To: "4010" ;tag=B5A00269-C426A3B4 CSeq: 105 NOTIFY Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK74dc7e63;rport (63) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=as7fcef0ca (53) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "4010" ;tag=B5A00269-C426A3B4 (65) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 105 NOTIFY (16) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 (49) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: presence (15) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:18] --- (10 headers 0 lines) --- [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #449 [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on 'ae75b12d-d9314ac3-30e8f7c6@172.16.16.135' of Request 105: Match Not Found [Feb 8 12:30:18] SIP Response message for INCOMING dialog NOTIFY arrived [Feb 8 12:30:18] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK74dc7e63;rport From: ;tag=as7fcef0ca To: "4010" ;tag=B5A00269-C426A3B4 CSeq: 105 NOTIFY Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK74dc7e63;rport (63) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=as7fcef0ca (53) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "4010" ;tag=B5A00269-C426A3B4 (65) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 105 NOTIFY (16) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: ae75b12d-d9314ac3-30e8f7c6@172.16.16.135 (49) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: presence (15) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:18] --- (10 headers 0 lines) --- [Feb 8 12:30:18] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on 'ae75b12d-d9314ac3-30e8f7c6@172.16.16.135' of Request 105: Match Found [Feb 8 12:30:18] SIP Response message for INCOMING dialog NOTIFY arrived [Feb 8 12:30:18] Really destroying SIP dialog '20bf7ba5561d16aa54293a26175a9ff7@172.16.16.12' Method: INVITE [Feb 8 12:30:18] Really destroying SIP dialog '7b13f17e349ea65f031be7f1482226e2@172.16.16.12' Method: BYE [Feb 8 12:30:20] DEBUG[12432]: chan_iax2.c:4775 raw_hangup: Raw Hangup 204.239.10.89:1696, src=0, dst=0 [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2488-chengb@172.16.16.133:5060 SIP/2.0 (50) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK21e7163a;rport (63) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as7ce2e79a (59) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (40) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 67dd8fd2705ff55d07a18a4e7229c2d8@172.16.16.12 (54) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:20 GMT (35) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:20] Reliably Transmitting (no NAT) to 172.16.16.133:5060: OPTIONS sip:2488-chengb@172.16.16.133:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK21e7163a;rport From: "asterisk" ;tag=as7ce2e79a To: Contact: Call-ID: 67dd8fd2705ff55d07a18a4e7229c2d8@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #450 [Feb 8 12:30:20] <--- SIP read from 172.16.16.133:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK21e7163a;rport From: "asterisk" ;tag=as7ce2e79a To: ;tag=24C8E851-200582FC CSeq: 102 OPTIONS Call-ID: 67dd8fd2705ff55d07a18a4e7229c2d8@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK21e7163a;rport (63) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as7ce2e79a (59) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=24C8E851-200582FC (62) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 67dd8fd2705ff55d07a18a4e7229c2d8@172.16.16.12 (54) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (45) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:20] --- (10 headers 0 lines) --- [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #450 [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '67dd8fd2705ff55d07a18a4e7229c2d8@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:20] Really destroying SIP dialog '67dd8fd2705ff55d07a18a4e7229c2d8@172.16.16.12' Method: OPTIONS [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2488-macdonap@172.16.16.135:5060 SIP/2.0 (52) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0adb391d;rport (63) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as597bff5b (59) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (42) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 19fa1ae63a58f42a0afab0d31f52c045@172.16.16.12 (54) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:20 GMT (35) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:20] Reliably Transmitting (no NAT) to 172.16.16.135:5060: OPTIONS sip:2488-macdonap@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0adb391d;rport From: "asterisk" ;tag=as597bff5b To: Contact: Call-ID: 19fa1ae63a58f42a0afab0d31f52c045@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #453 [Feb 8 12:30:20] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0adb391d;rport From: "asterisk" ;tag=as597bff5b To: ;tag=18247C52-36F88D49 CSeq: 102 OPTIONS Call-ID: 19fa1ae63a58f42a0afab0d31f52c045@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0adb391d;rport (63) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as597bff5b (59) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=18247C52-36F88D49 (64) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 19fa1ae63a58f42a0afab0d31f52c045@172.16.16.12 (54) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (47) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:20] --- (10 headers 0 lines) --- [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #453 [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '19fa1ae63a58f42a0afab0d31f52c045@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:20] Really destroying SIP dialog '19fa1ae63a58f42a0afab0d31f52c045@172.16.16.12' Method: OPTIONS [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2488-tessmanl@172.16.16.101:5060 SIP/2.0 (52) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK63803da3;rport (63) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as33e0fd16 (59) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (42) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 3072c43c16fa3f1c39a382890d2a7c0f@172.16.16.12 (54) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:20 GMT (35) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:20] Reliably Transmitting (no NAT) to 172.16.16.101:5060: OPTIONS sip:2488-tessmanl@172.16.16.101:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK63803da3;rport From: "asterisk" ;tag=as33e0fd16 To: Contact: Call-ID: 3072c43c16fa3f1c39a382890d2a7c0f@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #456 [Feb 8 12:30:20] <--- SIP read from 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK63803da3;rport From: "asterisk" ;tag=as33e0fd16 To: ;tag=738ED852-91EC526D CSeq: 102 OPTIONS Call-ID: 3072c43c16fa3f1c39a382890d2a7c0f@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK63803da3;rport (63) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as33e0fd16 (59) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=738ED852-91EC526D (64) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 3072c43c16fa3f1c39a382890d2a7c0f@172.16.16.12 (54) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (47) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:20] --- (10 headers 0 lines) --- [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #456 [Feb 8 12:30:20] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '3072c43c16fa3f1c39a382890d2a7c0f@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:20] Really destroying SIP dialog '3072c43c16fa3f1c39a382890d2a7c0f@172.16.16.12' Method: OPTIONS [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2225-upstairs@172.16.16.101:5060 SIP/2.0 (52) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK4aac2e76;rport (63) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as3770a826 (59) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (42) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 06a68f9f04718a6f41a6bf9a4df9e72a@172.16.16.12 (54) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:21 GMT (35) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:21] Reliably Transmitting (no NAT) to 172.16.16.101:5060: OPTIONS sip:2225-upstairs@172.16.16.101:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK4aac2e76;rport From: "asterisk" ;tag=as3770a826 To: Contact: Call-ID: 06a68f9f04718a6f41a6bf9a4df9e72a@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #459 [Feb 8 12:30:21] <--- SIP read from 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK4aac2e76;rport From: "asterisk" ;tag=as3770a826 To: ;tag=B7E1A301-9E0C409C CSeq: 102 OPTIONS Call-ID: 06a68f9f04718a6f41a6bf9a4df9e72a@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK4aac2e76;rport (63) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as3770a826 (59) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=B7E1A301-9E0C409C (64) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 06a68f9f04718a6f41a6bf9a4df9e72a@172.16.16.12 (54) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (47) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:21] --- (10 headers 0 lines) --- [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #459 [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '06a68f9f04718a6f41a6bf9a4df9e72a@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:21] Really destroying SIP dialog '06a68f9f04718a6f41a6bf9a4df9e72a@172.16.16.12' Method: OPTIONS [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2225-downstairs@172.16.16.135:5060 SIP/2.0 (54) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6928327e;rport (63) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as7ab7718f (59) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (44) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 278ab3e67bb3ee0c130e8e037354fb5a@172.16.16.12 (54) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:21 GMT (35) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:21] Reliably Transmitting (no NAT) to 172.16.16.135:5060: OPTIONS sip:2225-downstairs@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6928327e;rport From: "asterisk" ;tag=as7ab7718f To: Contact: Call-ID: 278ab3e67bb3ee0c130e8e037354fb5a@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #462 [Feb 8 12:30:21] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6928327e;rport From: "asterisk" ;tag=as7ab7718f To: ;tag=990CC2C5-555B8560 CSeq: 102 OPTIONS Call-ID: 278ab3e67bb3ee0c130e8e037354fb5a@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6928327e;rport (63) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as7ab7718f (59) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=990CC2C5-555B8560 (66) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 278ab3e67bb3ee0c130e8e037354fb5a@172.16.16.12 (54) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (49) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:21] --- (10 headers 0 lines) --- [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #462 [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '278ab3e67bb3ee0c130e8e037354fb5a@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:21] Really destroying SIP dialog '278ab3e67bb3ee0c130e8e037354fb5a@172.16.16.12' Method: OPTIONS [Feb 8 12:30:21] <--- SIP read from 172.16.16.104:5060 ---> REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 Via: SIP/2.0/UDP 172.16.16.104:5060;branch=z9hG4bKfa0e16469AF19225 From: "2312" ;tag=BD59DAB1-C5C40362 To: CSeq: 1547 REGISTER Call-ID: a1851f3d-f52ab6df-c9e54b8@172.16.16.104 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Authorization: Digest username="2312", realm="asterisk", nonce="1bc248cc", uri="sip:voip.dogmatix.dnv.org", response="bb709ef6234025caccfa161436e1dd6b", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 (42) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.104:5060;branch=z9hG4bKfa0e16469AF19225 (66) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "2312" ;tag=BD59DAB1-C5C40362 (67) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (36) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1547 REGISTER (19) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: a1851f3d-f52ab6df-c9e54b8@172.16.16.104 (48) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" (138) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Authorization: Digest username="2312", realm="asterisk", nonce="1bc248cc", uri="sip:voip.dogmatix.dnv.org", response="bb709ef6234025caccfa161436e1dd6b", algorithm=MD5 (166) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Expires: 3600 (13) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 0 (17) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:21] --- (12 headers 0 lines) --- [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for a1851f3d-f52ab6df-c9e54b8@172.16.16.104 - REGISTER (No RTP) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 8 12:30:21] Using latest REGISTER request as basis request [Feb 8 12:30:21] Sending to 172.16.16.104 : 5060 (no NAT) [Feb 8 12:30:21] <--- Transmitting (no NAT) to 172.16.16.104:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.104:5060;branch=z9hG4bKfa0e16469AF19225;received=172.16.16.104 From: "2312" ;tag=BD59DAB1-C5C40362 To: Call-ID: a1851f3d-f52ab6df-c9e54b8@172.16.16.104 CSeq: 1547 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:21] <--- Transmitting (no NAT) to 172.16.16.104:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.16.104:5060;branch=z9hG4bKfa0e16469AF19225;received=172.16.16.104 From: "2312" ;tag=BD59DAB1-C5C40362 To: ;tag=as52a9edc5 Call-ID: a1851f3d-f52ab6df-c9e54b8@172.16.16.104 CSeq: 1547 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c35935b" Content-Length: 0 <------------> [Feb 8 12:30:21] Scheduling destruction of SIP dialog 'a1851f3d-f52ab6df-c9e54b8@172.16.16.104' in 32000 ms (Method: REGISTER) [Feb 8 12:30:21] <--- SIP read from 172.16.16.104:5060 ---> REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 Via: SIP/2.0/UDP 172.16.16.104:5060;branch=z9hG4bKef0f8d409E4E3387 From: "2312" ;tag=BD59DAB1-C5C40362 To: CSeq: 1548 REGISTER Call-ID: a1851f3d-f52ab6df-c9e54b8@172.16.16.104 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Authorization: Digest username="2312", realm="asterisk", nonce="4c35935b", uri="sip:voip.dogmatix.dnv.org", response="f44ad41c62448b9bc75b4c8393beb63b", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 (42) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.104:5060;branch=z9hG4bKef0f8d409E4E3387 (66) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "2312" ;tag=BD59DAB1-C5C40362 (67) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (36) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1548 REGISTER (19) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: a1851f3d-f52ab6df-c9e54b8@172.16.16.104 (48) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" (138) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Authorization: Digest username="2312", realm="asterisk", nonce="4c35935b", uri="sip:voip.dogmatix.dnv.org", response="f44ad41c62448b9bc75b4c8393beb63b", algorithm=MD5 (166) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Expires: 3600 (13) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 0 (17) [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:21] --- (12 headers 0 lines) --- [Feb 8 12:30:21] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 8 12:30:21] Using latest REGISTER request as basis request [Feb 8 12:30:21] Sending to 172.16.16.104 : 5060 (no NAT) [Feb 8 12:30:21] <--- Transmitting (no NAT) to 172.16.16.104:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.104:5060;branch=z9hG4bKef0f8d409E4E3387;received=172.16.16.104 From: "2312" ;tag=BD59DAB1-C5C40362 To: Call-ID: a1851f3d-f52ab6df-c9e54b8@172.16.16.104 CSeq: 1548 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:21] -- Saved useragent "PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098" for peer 2312 [Feb 8 12:30:21] <--- Transmitting (no NAT) to 172.16.16.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.104:5060;branch=z9hG4bKef0f8d409E4E3387;received=172.16.16.104 From: "2312" ;tag=BD59DAB1-C5C40362 To: ;tag=as52a9edc5 Call-ID: a1851f3d-f52ab6df-c9e54b8@172.16.16.104 CSeq: 1548 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Date: Thu, 08 Feb 2007 20:30:21 GMT Content-Length: 0 <------------> [Feb 8 12:30:21] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/2312 [Feb 8 12:30:21] Scheduling destruction of SIP dialog 'a1851f3d-f52ab6df-c9e54b8@172.16.16.104' in 32000 ms (Method: REGISTER) [Feb 8 12:30:21] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 2312 [Feb 8 12:30:21] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 2312 [Feb 8 12:30:21] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/2312 - state 1 (Not in use) [Feb 8 12:30:21] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 2312 [Feb 8 12:30:21] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 2312 [Feb 8 12:30:21] DEBUG[12527]: app_queue.c:546 changethread: Device 'SIP/2312' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 8 12:30:24] <--- SIP read from 172.16.16.100:5060 ---> REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 Via: SIP/2.0/UDP 172.16.16.100:5060;branch=z9hG4bKe0a8f71bC3A85440 From: "2471" ;tag=B62844DF-49406A3E To: CSeq: 1547 REGISTER Call-ID: f632c773-36bf7f21-e472b178@172.16.16.100 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Authorization: Digest username="2471", realm="asterisk", nonce="0701b2fd", uri="sip:voip.dogmatix.dnv.org", response="beee9b61fe98241d444309818395d00c", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 (42) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.100:5060;branch=z9hG4bKe0a8f71bC3A85440 (66) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "2471" ;tag=B62844DF-49406A3E (67) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (36) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1547 REGISTER (19) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: f632c773-36bf7f21-e472b178@172.16.16.100 (49) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" (138) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Authorization: Digest username="2471", realm="asterisk", nonce="0701b2fd", uri="sip:voip.dogmatix.dnv.org", response="beee9b61fe98241d444309818395d00c", algorithm=MD5 (166) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Expires: 3600 (13) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 0 (17) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:24] --- (12 headers 0 lines) --- [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for f632c773-36bf7f21-e472b178@172.16.16.100 - REGISTER (No RTP) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 8 12:30:24] Using latest REGISTER request as basis request [Feb 8 12:30:24] Sending to 172.16.16.100 : 5060 (no NAT) [Feb 8 12:30:24] <--- Transmitting (no NAT) to 172.16.16.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.100:5060;branch=z9hG4bKe0a8f71bC3A85440;received=172.16.16.100 From: "2471" ;tag=B62844DF-49406A3E To: Call-ID: f632c773-36bf7f21-e472b178@172.16.16.100 CSeq: 1547 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:24] <--- Transmitting (no NAT) to 172.16.16.100:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.16.100:5060;branch=z9hG4bKe0a8f71bC3A85440;received=172.16.16.100 From: "2471" ;tag=B62844DF-49406A3E To: ;tag=as3cc745c6 Call-ID: f632c773-36bf7f21-e472b178@172.16.16.100 CSeq: 1547 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3044741a" Content-Length: 0 <------------> [Feb 8 12:30:24] Scheduling destruction of SIP dialog 'f632c773-36bf7f21-e472b178@172.16.16.100' in 32000 ms (Method: REGISTER) [Feb 8 12:30:24] <--- SIP read from 172.16.16.100:5060 ---> REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 Via: SIP/2.0/UDP 172.16.16.100:5060;branch=z9hG4bK57769b898F3A28C6 From: "2471" ;tag=B62844DF-49406A3E To: CSeq: 1548 REGISTER Call-ID: f632c773-36bf7f21-e472b178@172.16.16.100 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Authorization: Digest username="2471", realm="asterisk", nonce="3044741a", uri="sip:voip.dogmatix.dnv.org", response="d971af0c43379eb1997eaaa3afe368f9", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 (42) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.100:5060;branch=z9hG4bK57769b898F3A28C6 (66) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "2471" ;tag=B62844DF-49406A3E (67) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (36) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1548 REGISTER (19) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: f632c773-36bf7f21-e472b178@172.16.16.100 (49) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" (138) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Authorization: Digest username="2471", realm="asterisk", nonce="3044741a", uri="sip:voip.dogmatix.dnv.org", response="d971af0c43379eb1997eaaa3afe368f9", algorithm=MD5 (166) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Expires: 3600 (13) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 0 (17) [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:24] --- (12 headers 0 lines) --- [Feb 8 12:30:24] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 8 12:30:24] Using latest REGISTER request as basis request [Feb 8 12:30:24] Sending to 172.16.16.100 : 5060 (no NAT) [Feb 8 12:30:24] <--- Transmitting (no NAT) to 172.16.16.100:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.100:5060;branch=z9hG4bK57769b898F3A28C6;received=172.16.16.100 From: "2471" ;tag=B62844DF-49406A3E To: Call-ID: f632c773-36bf7f21-e472b178@172.16.16.100 CSeq: 1548 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:24] -- Saved useragent "PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098" for peer 2471 [Feb 8 12:30:24] <--- Transmitting (no NAT) to 172.16.16.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.100:5060;branch=z9hG4bK57769b898F3A28C6;received=172.16.16.100 From: "2471" ;tag=B62844DF-49406A3E To: ;tag=as3cc745c6 Call-ID: f632c773-36bf7f21-e472b178@172.16.16.100 CSeq: 1548 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Date: Thu, 08 Feb 2007 20:30:24 GMT Content-Length: 0 <------------> [Feb 8 12:30:24] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/2471 [Feb 8 12:30:24] Scheduling destruction of SIP dialog 'f632c773-36bf7f21-e472b178@172.16.16.100' in 32000 ms (Method: REGISTER) [Feb 8 12:30:24] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 2471 [Feb 8 12:30:24] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 2471 [Feb 8 12:30:24] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/2471 - state 1 (Not in use) [Feb 8 12:30:24] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 2471 [Feb 8 12:30:24] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 2471 [Feb 8 12:30:24] DEBUG[12528]: app_queue.c:546 changethread: Device 'SIP/2471' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:4012@172.16.16.133:5060 SIP/2.0 (43) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3b72b195;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as5c64990b (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 6891f58a2dfe01d415db91a6063b8704@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:29 GMT (35) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:29] Reliably Transmitting (no NAT) to 172.16.16.133:5060: OPTIONS sip:4012@172.16.16.133:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3b72b195;rport From: "asterisk" ;tag=as5c64990b To: Contact: Call-ID: 6891f58a2dfe01d415db91a6063b8704@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #471 [Feb 8 12:30:29] <--- SIP read from 172.16.16.133:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3b72b195;rport From: "asterisk" ;tag=as5c64990b To: ;tag=27043238-3B5FB93F CSeq: 102 OPTIONS Call-ID: 6891f58a2dfe01d415db91a6063b8704@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3b72b195;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as5c64990b (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=27043238-3B5FB93F (55) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 6891f58a2dfe01d415db91a6063b8704@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:29] --- (10 headers 0 lines) --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #471 [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '6891f58a2dfe01d415db91a6063b8704@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:4011@172.16.16.101:5060 SIP/2.0 (43) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK53d04cfc;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as59c8057a (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 201c32e67488d8f20e82f3d848dbe803@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:29 GMT (35) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:29] Reliably Transmitting (no NAT) to 172.16.16.101:5060: OPTIONS sip:4011@172.16.16.101:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK53d04cfc;rport From: "asterisk" ;tag=as59c8057a To: Contact: Call-ID: 201c32e67488d8f20e82f3d848dbe803@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #474 [Feb 8 12:30:29] <--- SIP read from 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK53d04cfc;rport From: "asterisk" ;tag=as59c8057a To: ;tag=FD7B8330-BDFC044B CSeq: 102 OPTIONS Call-ID: 201c32e67488d8f20e82f3d848dbe803@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK53d04cfc;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as59c8057a (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=FD7B8330-BDFC044B (55) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 201c32e67488d8f20e82f3d848dbe803@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:29] --- (10 headers 0 lines) --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #474 [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '201c32e67488d8f20e82f3d848dbe803@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) [Feb 8 12:30:29] Scheduling destruction of SIP dialog '2788885226ddedfa5750b2366e4af152@172.16.16.12' in 6400 ms (Method: NOTIFY) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: NOTIFY sip:2471@172.16.16.100:5060 SIP/2.0 (42) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK7329591b;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as7b3d78e3 (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 2788885226ddedfa5750b2366e4af152@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 NOTIFY (16) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Event: message-summary (22) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Content-Type: application/simple-message-summary (48) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 92 (18) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: Messages-Waiting: no (20) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: Message-Account: sip:asterisk@172.16.16.12 (42) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: Voice-Message: 0/0 (0/0) (24) [Feb 8 12:30:29] Reliably Transmitting (no NAT) to 172.16.16.100:5060: NOTIFY sip:2471@172.16.16.100:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK7329591b;rport From: "asterisk" ;tag=as7b3d78e3 To: Contact: Call-ID: 2788885226ddedfa5750b2366e4af152@172.16.16.12 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:asterisk@172.16.16.12 Voice-Message: 0/0 (0/0) --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #478 [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:4010@172.16.16.135:5060 SIP/2.0 (43) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6658fe89;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as3518efea (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 43f2ba7d76f3efa8369e73443d4502e3@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:29 GMT (35) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:29] Reliably Transmitting (no NAT) to 172.16.16.135:5060: OPTIONS sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6658fe89;rport From: "asterisk" ;tag=as3518efea To: Contact: Call-ID: 43f2ba7d76f3efa8369e73443d4502e3@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:29 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #479 [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) [Feb 8 12:30:29] Scheduling destruction of SIP dialog '510d280101bb49975328e6d37ab39ab9@172.16.16.12' in 6400 ms (Method: NOTIFY) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: NOTIFY sip:2312@172.16.16.104:5060 SIP/2.0 (42) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK035d4cec;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as51573416 (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 510d280101bb49975328e6d37ab39ab9@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 NOTIFY (16) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Event: message-summary (22) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Content-Type: application/simple-message-summary (48) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 92 (18) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: Messages-Waiting: no (20) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: Message-Account: sip:asterisk@172.16.16.12 (42) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: Voice-Message: 0/0 (0/0) (24) [Feb 8 12:30:29] Reliably Transmitting (no NAT) to 172.16.16.104:5060: NOTIFY sip:2312@172.16.16.104:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK035d4cec;rport From: "asterisk" ;tag=as51573416 To: Contact: Call-ID: 510d280101bb49975328e6d37ab39ab9@172.16.16.12 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:asterisk@172.16.16.12 Voice-Message: 0/0 (0/0) --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #482 [Feb 8 12:30:29] <--- SIP read from 172.16.16.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK7329591b;rport From: "asterisk" ;tag=as7b3d78e3 To: ;tag=C8E7DA9C-C9C01807 CSeq: 102 NOTIFY Call-ID: 2788885226ddedfa5750b2366e4af152@172.16.16.12 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK7329591b;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as7b3d78e3 (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=C8E7DA9C-C9C01807 (55) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 NOTIFY (16) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 2788885226ddedfa5750b2366e4af152@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: message-summary (22) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:29] --- (10 headers 0 lines) --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #478 [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '2788885226ddedfa5750b2366e4af152@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:29] Really destroying SIP dialog '2788885226ddedfa5750b2366e4af152@172.16.16.12' Method: NOTIFY [Feb 8 12:30:29] Really destroying SIP dialog '201c32e67488d8f20e82f3d848dbe803@172.16.16.12' Method: OPTIONS [Feb 8 12:30:29] Really destroying SIP dialog '6891f58a2dfe01d415db91a6063b8704@172.16.16.12' Method: OPTIONS [Feb 8 12:30:29] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6658fe89;rport From: "asterisk" ;tag=as3518efea To: ;tag=F3BF4AEC-8525E29B CSeq: 102 OPTIONS Call-ID: 43f2ba7d76f3efa8369e73443d4502e3@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6658fe89;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as3518efea (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=F3BF4AEC-8525E29B (55) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 43f2ba7d76f3efa8369e73443d4502e3@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:29] --- (10 headers 0 lines) --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #479 [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '43f2ba7d76f3efa8369e73443d4502e3@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:29] Really destroying SIP dialog '43f2ba7d76f3efa8369e73443d4502e3@172.16.16.12' Method: OPTIONS [Feb 8 12:30:29] <--- SIP read from 172.16.16.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK035d4cec;rport From: "asterisk" ;tag=as51573416 To: ;tag=63ADAB19-9940DCAA CSeq: 102 NOTIFY Call-ID: 510d280101bb49975328e6d37ab39ab9@172.16.16.12 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK035d4cec;rport (63) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as51573416 (59) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=63ADAB19-9940DCAA (55) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 NOTIFY (16) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 510d280101bb49975328e6d37ab39ab9@172.16.16.12 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: message-summary (22) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:29] --- (10 headers 0 lines) --- [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #482 [Feb 8 12:30:29] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '510d280101bb49975328e6d37ab39ab9@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:29] Really destroying SIP dialog '510d280101bb49975328e6d37ab39ab9@172.16.16.12' Method: NOTIFY [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2471@172.16.16.100:5060 SIP/2.0 (43) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK08cf18a9;rport (63) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as097616c8 (59) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 7f395dc25aed35c417b47b8b47f8efa5@172.16.16.12 (54) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:30 GMT (35) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:30] Reliably Transmitting (no NAT) to 172.16.16.100:5060: OPTIONS sip:2471@172.16.16.100:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK08cf18a9;rport From: "asterisk" ;tag=as097616c8 To: Contact: Call-ID: 7f395dc25aed35c417b47b8b47f8efa5@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #484 [Feb 8 12:30:30] <--- SIP read from 172.16.16.100:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK08cf18a9;rport From: "asterisk" ;tag=as097616c8 To: ;tag=AF0E6B33-8876D342 CSeq: 102 OPTIONS Call-ID: 7f395dc25aed35c417b47b8b47f8efa5@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK08cf18a9;rport (63) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as097616c8 (59) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=AF0E6B33-8876D342 (55) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 7f395dc25aed35c417b47b8b47f8efa5@172.16.16.12 (54) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:30] --- (10 headers 0 lines) --- [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #484 [Feb 8 12:30:30] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '7f395dc25aed35c417b47b8b47f8efa5@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:30] Really destroying SIP dialog '7f395dc25aed35c417b47b8b47f8efa5@172.16.16.12' Method: OPTIONS [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2420@172.16.16.119:5060 SIP/2.0 (43) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK63d9c977;rport (63) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as2f97e9f9 (59) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 0ed9e1e724b5738e2af0fa9238ce0be4@172.16.16.12 (54) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:31 GMT (35) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:31] Reliably Transmitting (no NAT) to 172.16.16.119:5060: OPTIONS sip:2420@172.16.16.119:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK63d9c977;rport From: "asterisk" ;tag=as2f97e9f9 To: Contact: Call-ID: 0ed9e1e724b5738e2af0fa9238ce0be4@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #487 [Feb 8 12:30:31] <--- SIP read from 172.16.16.119:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK63d9c977;rport From: "asterisk" ;tag=as2f97e9f9 To: ;tag=C3FBA6E6-C97788F9 CSeq: 102 OPTIONS Call-ID: 0ed9e1e724b5738e2af0fa9238ce0be4@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK63d9c977;rport (63) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as2f97e9f9 (59) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=C3FBA6E6-C97788F9 (55) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 0ed9e1e724b5738e2af0fa9238ce0be4@172.16.16.12 (54) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:31] --- (10 headers 0 lines) --- [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #487 [Feb 8 12:30:31] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '0ed9e1e724b5738e2af0fa9238ce0be4@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:31] Really destroying SIP dialog '0ed9e1e724b5738e2af0fa9238ce0be4@172.16.16.12' Method: OPTIONS [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2348-Polycom@172.16.16.132:5060 SIP/2.0 (51) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK642302ef;rport (63) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as6f3411d8 (59) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (41) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5861199966e40d722fbff57503968e4b@172.16.16.12 (54) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:32 GMT (35) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:32] Reliably Transmitting (no NAT) to 172.16.16.132:5060: OPTIONS sip:2348-Polycom@172.16.16.132:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK642302ef;rport From: "asterisk" ;tag=as6f3411d8 To: Contact: Call-ID: 5861199966e40d722fbff57503968e4b@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #490 [Feb 8 12:30:32] <--- SIP read from 172.16.16.132:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK642302ef;rport From: "asterisk" ;tag=as6f3411d8 To: ;tag=9015E25D-A07DEA46 CSeq: 102 OPTIONS Call-ID: 5861199966e40d722fbff57503968e4b@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK642302ef;rport (63) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as6f3411d8 (59) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=9015E25D-A07DEA46 (63) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5861199966e40d722fbff57503968e4b@172.16.16.12 (54) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (46) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.7.0098 (54) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:32] --- (10 headers 0 lines) --- [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #490 [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '5861199966e40d722fbff57503968e4b@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:32] Really destroying SIP dialog '5861199966e40d722fbff57503968e4b@172.16.16.12' Method: OPTIONS [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2312@172.16.16.104:5060 SIP/2.0 (43) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK4581371f;rport (63) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as34ed146e (59) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 11d95008692a023e5774f3907f63676c@172.16.16.12 (54) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:30:32 GMT (35) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:30:32] Reliably Transmitting (no NAT) to 172.16.16.104:5060: OPTIONS sip:2312@172.16.16.104:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK4581371f;rport From: "asterisk" ;tag=as34ed146e To: Contact: Call-ID: 11d95008692a023e5774f3907f63676c@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:30:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #493 [Feb 8 12:30:32] <--- SIP read from 172.16.16.104:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK4581371f;rport From: "asterisk" ;tag=as34ed146e To: ;tag=2FC80ACE-F511855B CSeq: 102 OPTIONS Call-ID: 11d95008692a023e5774f3907f63676c@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK4581371f;rport (63) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as34ed146e (59) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=2FC80ACE-F511855B (55) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 11d95008692a023e5774f3907f63676c@172.16.16.12 (54) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:32] --- (10 headers 0 lines) --- [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #493 [Feb 8 12:30:32] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '11d95008692a023e5774f3907f63676c@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:32] Really destroying SIP dialog '11d95008692a023e5774f3907f63676c@172.16.16.12' Method: OPTIONS [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:2009 __sip_autodestruct: Auto destroying SIP dialog '5588e51e-c415a168-c97a9e83@172.16.16.101' [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:2016 __sip_autodestruct: Finally hanging up channel after transfer: 5588e51e-c415a168-c97a9e83@172.16.16.101 [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:5622 reqprep: Strict routing enforced for session 5588e51e-c415a168-c97a9e83@172.16.16.101 [Feb 8 12:30:44] set_destination: Parsing for address/port to send to [Feb 8 12:30:44] set_destination: set destination to 172.16.16.101, port 5060 [Feb 8 12:30:44] Reliably Transmitting (no NAT) to 172.16.16.101:5060: BYE sip:4011@172.16.16.101:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0f1b9144;rport From: ;tag=as0083934c To: "4011" ;tag=225C83B2-C4872DCD Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #496 [Feb 8 12:30:44] Scheduling destruction of SIP dialog '5588e51e-c415a168-c97a9e83@172.16.16.101' in 32000 ms (Method: BYE) [Feb 8 12:30:44] <--- SIP read from 172.16.16.101:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0f1b9144;rport From: ;tag=as0083934c To: "4011" ;tag=225C83B2-C4872DCD CSeq: 103 BYE Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 481 Call Leg/Transaction Does Not Exist (47) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0f1b9144;rport (63) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=as0083934c (64) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "4011" ;tag=225C83B2-C4872DCD (65) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 103 BYE (13) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 5588e51e-c415a168-c97a9e83@172.16.16.101 (49) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Content-Length: 0 (17) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: (0) [Feb 8 12:30:44] --- (8 headers 0 lines) --- [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #496 [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '5588e51e-c415a168-c97a9e83@172.16.16.101' of Request 103: Match Not Found [Feb 8 12:30:44] SIP Response message for INCOMING dialog BYE arrived [Feb 8 12:30:44] Really destroying SIP dialog '5588e51e-c415a168-c97a9e83@172.16.16.101' Method: BYE [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:3002 update_call_counter: Updating call counter for incoming call [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:3050 update_call_counter: Call from peer '4011' removed from call limit 200 [Feb 8 12:30:44] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/4011 [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:2893 __sip_destroy: This call did not properly clean up call limits. Call ID 5588e51e-c415a168-c97a9e83@172.16.16.101 [Feb 8 12:30:44] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:44] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:44] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/4011 - state 1 (Not in use) [Feb 8 12:30:44] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:44] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:44] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 4011 [Feb 8 12:30:44] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 4011 [Feb 8 12:30:44] Reliably Transmitting (no NAT) to 172.16.16.135:5060: NOTIFY sip:4010@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0138c944;rport From: ;tag=as1a9ebeca To: "4010" ;tag=D2C77ADC-9D78A5CB Contact: Call-ID: e9654c50-a338ad2e-417f8575@172.16.16.135 CSeq: 105 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 362
--- [Feb 8 12:30:44] DEBUG[12424]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #498 [Feb 8 12:30:44] Extension Changed 4011 new state Idle for Notify User 4010 [Feb 8 12:30:44] DEBUG[12532]: app_queue.c:546 changethread: Device 'SIP/4011' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 8 12:30:44] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0138c944;rport From: ;tag=as1a9ebeca To: "4010" ;tag=D2C77ADC-9D78A5CB CSeq: 105 NOTIFY Call-ID: e9654c50-a338ad2e-417f8575@172.16.16.135 Contact: Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK0138c944;rport (63) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: ;tag=as1a9ebeca (53) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: "4010" ;tag=D2C77ADC-9D78A5CB (65) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 105 NOTIFY (16) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: e9654c50-a338ad2e-417f8575@172.16.16.135 (49) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: presence (15) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:44] --- (10 headers 0 lines) --- [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #498 [Feb 8 12:30:44] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on 'e9654c50-a338ad2e-417f8575@172.16.16.135' of Request 105: Match Not Found [Feb 8 12:30:44] SIP Response message for INCOMING dialog NOTIFY arrived [Feb 8 12:30:50] DEBUG[12433]: chan_iax2.c:4775 raw_hangup: Raw Hangup 204.239.10.89:1758, src=0, dst=0 [Feb 8 12:30:51] <--- SIP read from 172.16.16.119:5060 ---> REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 Via: SIP/2.0/UDP 172.16.16.119:5060;branch=z9hG4bKcd0ee79bE4DC7058 From: "SHAYNE DUNLOP" ;tag=7B12EE0F-9247413A To: CSeq: 1547 REGISTER Call-ID: d487be6b-15912c2d-ce7833a8@172.16.16.119 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Authorization: Digest username="2420", realm="asterisk", nonce="596bfd59", uri="sip:voip.dogmatix.dnv.org", response="f9d1469ae4596ad35e4673860ecd19eb", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 (42) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.119:5060;branch=z9hG4bKcd0ee79bE4DC7058 (66) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "SHAYNE DUNLOP" ;tag=7B12EE0F-9247413A (76) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (36) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1547 REGISTER (19) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: d487be6b-15912c2d-ce7833a8@172.16.16.119 (49) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" (138) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Authorization: Digest username="2420", realm="asterisk", nonce="596bfd59", uri="sip:voip.dogmatix.dnv.org", response="f9d1469ae4596ad35e4673860ecd19eb", algorithm=MD5 (166) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Expires: 3600 (13) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 0 (17) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:51] --- (12 headers 0 lines) --- [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for d487be6b-15912c2d-ce7833a8@172.16.16.119 - REGISTER (No RTP) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 8 12:30:51] Using latest REGISTER request as basis request [Feb 8 12:30:51] Sending to 172.16.16.119 : 5060 (no NAT) [Feb 8 12:30:51] <--- Transmitting (no NAT) to 172.16.16.119:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.119:5060;branch=z9hG4bKcd0ee79bE4DC7058;received=172.16.16.119 From: "SHAYNE DUNLOP" ;tag=7B12EE0F-9247413A To: Call-ID: d487be6b-15912c2d-ce7833a8@172.16.16.119 CSeq: 1547 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:51] <--- Transmitting (no NAT) to 172.16.16.119:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.16.119:5060;branch=z9hG4bKcd0ee79bE4DC7058;received=172.16.16.119 From: "SHAYNE DUNLOP" ;tag=7B12EE0F-9247413A To: ;tag=as785fba72 Call-ID: d487be6b-15912c2d-ce7833a8@172.16.16.119 CSeq: 1547 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c5274ef" Content-Length: 0 <------------> [Feb 8 12:30:51] Scheduling destruction of SIP dialog 'd487be6b-15912c2d-ce7833a8@172.16.16.119' in 32000 ms (Method: REGISTER) [Feb 8 12:30:51] <--- SIP read from 172.16.16.119:5060 ---> REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 Via: SIP/2.0/UDP 172.16.16.119:5060;branch=z9hG4bK3f2285d12E888EA From: "SHAYNE DUNLOP" ;tag=7B12EE0F-9247413A To: CSeq: 1548 REGISTER Call-ID: d487be6b-15912c2d-ce7833a8@172.16.16.119 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Authorization: Digest username="2420", realm="asterisk", nonce="3c5274ef", uri="sip:voip.dogmatix.dnv.org", response="6292f9b50737b910314b34b18bab4ad2", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: REGISTER sip:voip.dogmatix.dnv.org SIP/2.0 (42) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.119:5060;branch=z9hG4bK3f2285d12E888EA (65) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "SHAYNE DUNLOP" ;tag=7B12EE0F-9247413A (76) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (36) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 1548 REGISTER (19) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: d487be6b-15912c2d-ce7833a8@172.16.16.119 (49) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" (138) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Authorization: Digest username="2420", realm="asterisk", nonce="3c5274ef", uri="sip:voip.dogmatix.dnv.org", response="6292f9b50737b910314b34b18bab4ad2", algorithm=MD5 (166) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Max-Forwards: 70 (16) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Expires: 3600 (13) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 0 (17) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:51] --- (12 headers 0 lines) --- [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:14575 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Feb 8 12:30:51] Using latest REGISTER request as basis request [Feb 8 12:30:51] Sending to 172.16.16.119 : 5060 (no NAT) [Feb 8 12:30:51] <--- Transmitting (no NAT) to 172.16.16.119:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.119:5060;branch=z9hG4bK3f2285d12E888EA;received=172.16.16.119 From: "SHAYNE DUNLOP" ;tag=7B12EE0F-9247413A To: Call-ID: d487be6b-15912c2d-ce7833a8@172.16.16.119 CSeq: 1548 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Feb 8 12:30:51] -- Saved useragent "PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098" for peer 2420 [Feb 8 12:30:51] <--- Transmitting (no NAT) to 172.16.16.119:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.119:5060;branch=z9hG4bK3f2285d12E888EA;received=172.16.16.119 From: "SHAYNE DUNLOP" ;tag=7B12EE0F-9247413A To: ;tag=as785fba72 Call-ID: d487be6b-15912c2d-ce7833a8@172.16.16.119 CSeq: 1548 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Date: Thu, 08 Feb 2007 20:30:51 GMT Content-Length: 0 <------------> [Feb 8 12:30:51] DEBUG[12438]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/2420 [Feb 8 12:30:51] Scheduling destruction of SIP dialog 'd487be6b-15912c2d-ce7833a8@172.16.16.119' in 32000 ms (Method: REGISTER) [Feb 8 12:30:51] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 2420 [Feb 8 12:30:51] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 2420 [Feb 8 12:30:51] DEBUG[12424]: devicestate.c:287 do_state_change: Changing state for SIP/2420 - state 1 (Not in use) [Feb 8 12:30:51] DEBUG[12424]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 2420 [Feb 8 12:30:51] DEBUG[12424]: chan_sip.c:15180 sip_devicestate: Checking device state for peer 2420 [Feb 8 12:30:51] DEBUG[12533]: app_queue.c:546 changethread: Device 'SIP/2420' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) [Feb 8 12:30:51] Scheduling destruction of SIP dialog '28b511984c03bd886a09db0708dc6b71@172.16.16.12' in 6400 ms (Method: NOTIFY) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: NOTIFY sip:2420@172.16.16.119:5060 SIP/2.0 (42) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK33466497;rport (63) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as69e4d7db (59) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (33) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 28b511984c03bd886a09db0708dc6b71@172.16.16.12 (54) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 NOTIFY (16) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Event: message-summary (22) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Content-Type: application/simple-message-summary (48) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Content-Length: 92 (18) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: (0) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: Messages-Waiting: no (20) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: Message-Account: sip:asterisk@172.16.16.12 (42) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4594 parse_request: Line: Voice-Message: 0/0 (0/0) (24) [Feb 8 12:30:51] Reliably Transmitting (no NAT) to 172.16.16.119:5060: NOTIFY sip:2420@172.16.16.119:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK33466497;rport From: "asterisk" ;tag=as69e4d7db To: Contact: Call-ID: 28b511984c03bd886a09db0708dc6b71@172.16.16.12 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:asterisk@172.16.16.12 Voice-Message: 0/0 (0/0) --- [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #503 [Feb 8 12:30:51] <--- SIP read from 172.16.16.119:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK33466497;rport From: "asterisk" ;tag=as69e4d7db To: ;tag=A5E0629C-C65EED3F CSeq: 102 NOTIFY Call-ID: 28b511984c03bd886a09db0708dc6b71@172.16.16.12 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK33466497;rport (63) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as69e4d7db (59) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=A5E0629C-C65EED3F (55) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 NOTIFY (16) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 28b511984c03bd886a09db0708dc6b71@172.16.16.12 (54) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (38) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Event: message-summary (22) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:30:51] --- (10 headers 0 lines) --- [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #503 [Feb 8 12:30:51] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '28b511984c03bd886a09db0708dc6b71@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:30:51] Really destroying SIP dialog '28b511984c03bd886a09db0708dc6b71@172.16.16.12' Method: NOTIFY sip show inusehow hints[Feb 8 12:30:53] DEBUG[12438]: chan_sip.c:2009 __sip_autodestruct: Auto destroying SIP dialog 'a1851f3d-f52ab6df-c9e54b8@172.16.16.104' [Feb 8 12:30:53] DEBUG[12438]: chan_sip.c:3108 sip_destroy: Destroying SIP dialog a1851f3d-f52ab6df-c9e54b8@172.16.16.104 [Feb 8 12:30:53] Really destroying SIP dialog 'a1851f3d-f52ab6df-c9e54b8@172.16.16.104' Method: REGISTER -= Registered Asterisk Dial Plan Hints =- 8716@private-extensions : SIP/8716 State:Unavailable Watchers 0 8715@private-extensions : SIP/8715 State:Unavailable Watchers 0 8714@private-extensions : SIP/8714 State:Unavailable Watchers 0 8713@private-extensions : SIP/8713 State:Unavailable Watchers 0 8712@private-extensions : SIP/8712 State:Unavailable Watchers 0 8711@private-extensions : SIP/8711 State:Unavailable Watchers 0 8710@private-extensions : SIP/8710 State:Unavailable Watchers 0 8696@private-extensions : SIP/8696 State:Unavailable Watchers 0 8695@private-extensions : SIP/8695 State:Unavailable Watchers 0 8694@private-extensions : SIP/8694 State:Unavailable Watchers 0 8693@private-extensions : SIP/8693 State:Unavailable Watchers 0 8692@private-extensions : SIP/8692 State:Unavailable Watchers 0 8691@private-extensions : SIP/8691 State:Unavailable Watchers 0 8690@private-extensions : SIP/8690 State:Unavailable Watchers 0 8169@private-extensions : SIP/8169 State:Unavailable Watchers 0 8168@private-extensions : SIP/8168 State:Unavailable Watchers 0 8166@private-extensions : SIP/8166 State:Unavailable Watchers 0 8162@private-extensions : SIP/8162 State:Unavailable Watchers 0 8161@private-extensions : SIP/8161 State:Unavailable Watchers 0 8160@private-extensions : SIP/8160 State:Unavailable Watchers 0 8159@private-extensions : SIP/8159 State:Unavailable Watchers 0 8158@private-extensions : SIP/8158 State:Unavailable Watchers 0 8157@private-extensions : SIP/8157 State:Unavailable Watchers 0 8156@private-extensions : SIP/8156 State:Unavailable Watchers 0 8155@private-extensions : SIP/8155 State:Unavailable Watchers 0 8154@private-extensions : SIP/8154 State:Unavailable Watchers 0 8153@private-extensions : SIP/8153 State:Unavailable Watchers 0 8152@private-extensions : SIP/8152 State:Unavailable Watchers 0 8151@private-extensions : SIP/8151 State:Unavailable Watchers 0 8150@private-extensions : SIP/8150 State:Unavailable Watchers 0 8016@private-extensions : SIP/8016 State:Unavailable Watchers 0 8015@private-extensions : SIP/8015 State:Unavailable Watchers 0 8014@private-extensions : SIP/8014 State:Unavailable Watchers 0 8013@private-extensions : SIP/8013 State:Unavailable Watchers 0 8012@private-extensions : SIP/8012 State:Unavailable Watchers 0 8011@private-extensions : SIP/8011 State:Unavailable Watchers 0 8010@private-extensions : SIP/8010 State:Unavailable Watchers 0 8009@private-extensions : SIP/8009 State:Unavailable Watchers 0 8008@private-extensions : SIP/8008 State:Unavailable Watchers 0 8007@private-extensions : SIP/8007 State:Unavailable Watchers 0 8006@private-extensions : SIP/8006 State:Unavailable Watchers 0 8005@private-extensions : SIP/8005 State:Unavailable Watchers 0 8004@private-extensions : SIP/8004 State:Unavailable Watchers 0 8003@private-extensions : SIP/8003 State:Unavailable Watchers 0 8002@private-extensions : SIP/8002 State:Unavailable Watchers 0 8001@private-extensions : SIP/8001 State:Unavailable Watchers 0 4012@private-extensions : SIP/4012 State:Idle Watchers 1 4011@private-extensions : SIP/4011 State:Idle Watchers 1 4010@private-extensions : SIP/4010 State:Idle Watchers 0 4009@private-extensions : SIP/4009 State:Unavailable Watchers 0 4008@private-extensions : SIP/4008 State:Unavailable Watchers 0 4007@private-extensions : SIP/4007 State:Unavailable Watchers 0 4006@private-extensions : SIP/4006 State:Unavailable Watchers 0 4005@private-extensions : SIP/4005 State:Unavailable Watchers 0 4004@private-extensions : SIP/4004 State:Unavailable Watchers 0 4003@private-extensions : SIP/4003 State:Unavailable Watchers 0 4002@private-extensions : SIP/4002 State:Unavailable Watchers 0 4001@private-extensions : SIP/4001 State:Unavailable Watchers 0 3844@public-extensions : SIP/3844 State:Unavailable Watchers 0 3842@public-extensions : SIP/3842 State:Unavailable Watchers 0 3838@public-extensions : SIP/3838 State:Unavailable Watchers 0 3727@public-extensions : SIP/3727 State:Unavailable Watchers 0 3712@public-extensions : SIP/3712 State:Unavailable Watchers 0 3711@public-extensions : SIP/3711 State:Unavailable Watchers 0 3700@public-extensions : SIP/3700 State:Unavailable Watchers 0 3690@public-extensions : SIP/3690 State:Unavailable Watchers 0 3681@public-extensions : SIP/3681 State:Unavailable Watchers 0 3675@public-extensions : SIP/3675 State:Unavailable Watchers 0 3674@public-extensions : SIP/3674 State:Unavailable Watchers 0 3673@public-extensions : SIP/3673 State:Unavailable Watchers 0 3672@public-extensions : SIP/3672 State:Unavailable Watchers 0 3671@public-extensions : SIP/3671 State:Unavailable Watchers 0 3670@public-extensions : SIP/3670 State:Unavailable Watchers 0 3668@public-extensions : SIP/3668 State:Unavailable Watchers 0 3667@public-extensions : SIP/3667 State:Unavailable Watchers 0 3665@public-extensions : SIP/3665 State:Unavailable Watchers 0 3664@public-extensions : SIP/3664 State:Unavailable Watchers 0 3663@public-extensions : SIP/3663 State:Unavailable Watchers 0 3662@public-extensions : SIP/3662 State:Unavailable Watchers 0 3661@public-extensions : SIP/3661 State:Unavailable Watchers 0 3660@public-extensions : SIP/3660 State:Unavailable Watchers 0 3658@public-extensions : SIP/3658 State:Unavailable Watchers 0 3655@public-extensions : SIP/3655 State:Unavailable Watchers 0 3654@public-extensions : SIP/3654 State:Unavailable Watchers 0 3653@public-extensions : SIP/3653 State:Unavailable Watchers 0 3652@public-extensions : SIP/3652 State:Unavailable Watchers 0 3651@public-extensions : SIP/3651 State:Unavailable Watchers 0 3031@public-extensions : SIP/3031 State:Unavailable Watchers 0 3030@public-extensions : SIP/3030 State:Unavailable Watchers 0 2499@public-extensions : SIP/2499 State:Unavailable Watchers 0 2488@public-extensions : SIP/2488 State:Unavailable Watchers 0 2485@public-extensions : SIP/2485 State:Unavailable Watchers 0 2478@public-extensions : SIP/2478 State:Unavailable Watchers 0 2472@public-extensions : SIP/2472 State:Unavailable Watchers 0 2471@public-extensions : SIP/2471 State:Idle Watchers 0 2470@public-extensions : SIP/2470 State:Unavailable Watchers 0 2469@public-extensions : SIP/2469 State:Unavailable Watchers 0 2468@public-extensions : SIP/2468 State:Unavailable Watchers 0 2457@public-extensions : SIP/2457 State:Unavailable Watchers 0 2456@public-extensions : SIP/2456 State:Unavailable Watchers 0 2448@public-extensions : SIP/2448 State:Unavailable Watchers 0 2440@public-extensions : SIP/2440 State:Unavailable Watchers 0 2439@public-extensions : SIP/2439 State:Unavailable Watchers 0 2438@public-extensions : SIP/2438 State:Unavailable Watchers 0 2426@public-extensions : SIP/2426 State:Unavailable Watchers 0 2420@public-extensions : SIP/2420 State:Idle Watchers 0 2412@public-extensions : SIP/2412 State:Unavailable Watchers 0 2397@public-extensions : SIP/2397 State:Unavailable Watchers 0 2390@public-extensions : SIP/2390 State:Unavailable Watchers 0 2382@public-extensions : SIP/2382 State:Unavailable Watchers 0 2381@public-extensions : SIP/2381 State:Unavailable Watchers 0 2380@public-extensions : SIP/2380 State:Unavailable Watchers 0 2372@public-extensions : SIP/2372 State:Unavailable Watchers 0 2348@public-extensions : SIP/2348-Polycom State:Unavailable Watchers 0 2331@public-extensions : SIP/2331 State:Unavailable Watchers 0 2323@public-extensions : SIP/2323 State:Unavailable Watchers 0 2313@public-extensions : SIP/2313 State:Unavailable Watchers 0 2312@public-extensions : SIP/2312 State:Idle Watchers 0 2309@public-extensions : SIP/2309 State:Unavailable Watchers 0 2308@public-extensions : SIP/2308 State:Unavailable Watchers 0 2306@public-extensions : SIP/2306-Polycom&SIP State:Unavailable Watchers 0 2305@public-extensions : SIP/2305 State:Unavailable Watchers 0 2304@public-extensions : SIP/2304 State:Unavailable Watchers 0 2303@public-extensions : SIP/2303 State:Unavailable Watchers 0 2301@public-extensions : SIP/2301 State:Unavailable Watchers 0 2295@public-extensions : SIP/2295 State:Unavailable Watchers 0 2290@public-extensions : SIP/2290-ridgej State:Unavailable Watchers 0 2266@public-extensions : SIP/2266 State:Unavailable Watchers 0 2238@public-extensions : SIP/2238 State:Unavailable Watchers 0 2235@public-extensions : SIP/2235 State:Unavailable Watchers 0 2230@public-extensions : SIP/2230 State:Unavailable Watchers 0 2229@public-extensions : SIP/2229 State:Unavailable Watchers 0 2228@public-extensions : SIP/2228 State:Unavailable Watchers 0 2226@public-extensions : SIP/2226 State:Unavailable Watchers 0 2225@public-extensions : SIP/2225 State:Unavailable Watchers 0 2218@public-extensions : SIP/2218 State:Unavailable Watchers 0 2209@public-extensions : SIP/2206-mitchells&S State:Unavailable Watchers 0 2206@public-extensions : SIP/2206-ridgej State:Unavailable Watchers 0 ---------------- - 138 hints registered *CLI> show hintsip show inuse * User name In use Limit 4012 0 200 4011 0 200 4010 0 200 2348-Polycom 0 200 * Peer name In use Limit 4012 0/0 200 4011 0/0 200 4010 0/0 200 2348-Polycom 0/0 200 *CLI> [Feb 8 12:30:56] DEBUG[12438]: chan_sip.c:2009 __sip_autodestruct: Auto destroying SIP dialog 'f632c773-36bf7f21-e472b178@172.16.16.100' [Feb 8 12:30:56] DEBUG[12438]: chan_sip.c:3108 sip_destroy: Destroying SIP dialog f632c773-36bf7f21-e472b178@172.16.16.100 [Feb 8 12:30:56] Really destroying SIP dialog 'f632c773-36bf7f21-e472b178@172.16.16.100' Method: REGISTER [Feb 8 12:31:11] DEBUG[12428]: chan_iax2.c:7099 socket_process: Peer gabriola-out: got pong, lastms 16, historicms 16, maxms 2000 stop [Feb 8 12:31:20] DEBUG[12430]: chan_iax2.c:4775 raw_hangup: Raw Hangup 204.239.10.89:1820, src=0, dst=0 wh[Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2488-chengb@172.16.16.133:5060 SIP/2.0 (50) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK5f7b4b34;rport (63) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as5ba4dd09 (59) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (40) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 262a104673fbd37b5455b01d07597a84@172.16.16.12 (54) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:31:20 GMT (35) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:31:20] Reliably Transmitting (no NAT) to 172.16.16.133:5060: OPTIONS sip:2488-chengb@172.16.16.133:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK5f7b4b34;rport From: "asterisk" ;tag=as5ba4dd09 To: Contact: Call-ID: 262a104673fbd37b5455b01d07597a84@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:31:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #504 [Feb 8 12:31:20] <--- SIP read from 172.16.16.133:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK5f7b4b34;rport From: "asterisk" ;tag=as5ba4dd09 To: ;tag=BCEE452B-CA3EFBBE CSeq: 102 OPTIONS Call-ID: 262a104673fbd37b5455b01d07597a84@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK5f7b4b34;rport (63) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as5ba4dd09 (59) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=BCEE452B-CA3EFBBE (62) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 262a104673fbd37b5455b01d07597a84@172.16.16.12 (54) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (45) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:31:20] --- (10 headers 0 lines) --- [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #504 [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '262a104673fbd37b5455b01d07597a84@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:31:20] Really destroying SIP dialog '262a104673fbd37b5455b01d07597a84@172.16.16.12' Method: OPTIONS e[Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2488-macdonap@172.16.16.135:5060 SIP/2.0 (52) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK62da1ec3;rport (63) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as79348996 (59) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (42) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 4b80af2d1b5a644413a44ecd74802ee1@172.16.16.12 (54) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:31:20 GMT (35) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:31:20] Reliably Transmitting (no NAT) to 172.16.16.135:5060: OPTIONS sip:2488-macdonap@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK62da1ec3;rport From: "asterisk" ;tag=as79348996 To: Contact: Call-ID: 4b80af2d1b5a644413a44ecd74802ee1@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:31:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #507 n[Feb 8 12:31:20] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK62da1ec3;rport From: "asterisk" ;tag=as79348996 To: ;tag=89905FF7-ECA8BA6A CSeq: 102 OPTIONS Call-ID: 4b80af2d1b5a644413a44ecd74802ee1@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK62da1ec3;rport (63) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as79348996 (59) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=89905FF7-ECA8BA6A (64) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 4b80af2d1b5a644413a44ecd74802ee1@172.16.16.12 (54) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (47) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:31:20] --- (10 headers 0 lines) --- [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #507 [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '4b80af2d1b5a644413a44ecd74802ee1@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:31:20] Really destroying SIP dialog '4b80af2d1b5a644413a44ecd74802ee1@172.16.16.12' Method: OPTIONS [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2488-tessmanl@172.16.16.101:5060 SIP/2.0 (52) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3394e591;rport (63) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as53007ee8 (59) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (42) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 6bbb9fbf1ba5f9c45f620786433a482d@172.16.16.12 (54) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:31:20 GMT (35) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:31:20] Reliably Transmitting (no NAT) to 172.16.16.101:5060: OPTIONS sip:2488-tessmanl@172.16.16.101:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3394e591;rport From: "asterisk" ;tag=as53007ee8 To: Contact: Call-ID: 6bbb9fbf1ba5f9c45f620786433a482d@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:31:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:31:20] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #510 [Feb 8 12:31:21] <--- SIP read from 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3394e591;rport From: "asterisk" ;tag=as53007ee8 To: ;tag=510238DF-8C415D7A CSeq: 102 OPTIONS Call-ID: 6bbb9fbf1ba5f9c45f620786433a482d@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK3394e591;rport (63) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as53007ee8 (59) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=510238DF-8C415D7A (64) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 6bbb9fbf1ba5f9c45f620786433a482d@172.16.16.12 (54) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (47) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:31:21] --- (10 headers 0 lines) --- [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #510 [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '6bbb9fbf1ba5f9c45f620786433a482d@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:31:21] Really destroying SIP dialog '6bbb9fbf1ba5f9c45f620786433a482d@172.16.16.12' Method: OPTIONS [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2225-upstairs@172.16.16.101:5060 SIP/2.0 (52) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6328f40c;rport (63) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as323f64ae (59) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (42) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 7f3e14c021100f636482e91f4f7e4b2d@172.16.16.12 (54) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:31:21 GMT (35) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:31:21] Reliably Transmitting (no NAT) to 172.16.16.101:5060: OPTIONS sip:2225-upstairs@172.16.16.101:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6328f40c;rport From: "asterisk" ;tag=as323f64ae To: Contact: Call-ID: 7f3e14c021100f636482e91f4f7e4b2d@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:31:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #513 [Feb 8 12:31:21] <--- SIP read from 172.16.16.101:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6328f40c;rport From: "asterisk" ;tag=as323f64ae To: ;tag=B57B840E-49C20C29 CSeq: 102 OPTIONS Call-ID: 7f3e14c021100f636482e91f4f7e4b2d@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK6328f40c;rport (63) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as323f64ae (59) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=B57B840E-49C20C29 (64) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 7f3e14c021100f636482e91f4f7e4b2d@172.16.16.12 (54) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (47) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 (54) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:31:21] --- (10 headers 0 lines) --- [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #513 [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '7f3e14c021100f636482e91f4f7e4b2d@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:31:21] Really destroying SIP dialog '7f3e14c021100f636482e91f4f7e4b2d@172.16.16.12' Method: OPTIONS [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4299 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: OPTIONS sip:2225-downstairs@172.16.16.135:5060 SIP/2.0 (54) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK44085daf;rport (63) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as21f69103 (59) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: (44) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: Contact: (36) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 3cac5508350c0f7a715e846924a4e901@172.16.16.12 (54) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: CSeq: 102 OPTIONS (17) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: User-Agent: Asterisk PBX (24) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: Max-Forwards: 70 (16) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Date: Thu, 08 Feb 2007 20:31:21 GMT (35) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 11: Supported: replaces (19) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 12: Content-Length: 0 (17) [Feb 8 12:31:21] Reliably Transmitting (no NAT) to 172.16.16.135:5060: OPTIONS sip:2225-downstairs@172.16.16.135:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK44085daf;rport From: "asterisk" ;tag=as21f69103 To: Contact: Call-ID: 3cac5508350c0f7a715e846924a4e901@172.16.16.12 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 08 Feb 2007 20:31:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:1974 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #516 [Feb 8 12:31:21] <--- SIP read from 172.16.16.135:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK44085daf;rport From: "asterisk" ;tag=as21f69103 To: ;tag=EDC2D856-DAA7247D CSeq: 102 OPTIONS Call-ID: 3cac5508350c0f7a715e846924a4e901@172.16.16.12 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 Content-Length: 0 <-------------> [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 0: SIP/2.0 200 OK (14) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.16.12:5060;branch=z9hG4bK44085daf;rport (63) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 2: From: "asterisk" ;tag=as21f69103 (59) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 3: To: ;tag=EDC2D856-DAA7247D (66) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 4: CSeq: 102 OPTIONS (17) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 5: Call-ID: 3cac5508350c0f7a715e846924a4e901@172.16.16.12 (54) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 6: Contact: (49) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 7: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER (96) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 8: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.7.0098 (54) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 9: Content-Length: 0 (17) [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:4562 parse_request: Header 10: (0) [Feb 8 12:31:21] --- (10 headers 0 lines) --- [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:2078 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #516 [Feb 8 12:31:21] DEBUG[12438]: chan_sip.c:2088 __sip_ack: Stopping retransmission on '3cac5508350c0f7a715e846924a4e901@172.16.16.12' of Request 102: Match Not Found [Feb 8 12:31:21] Really destroying SIP dialog '3cac5508350c0f7a715e846924a4e901@172.16.16.12' Method: OPTIONS convenient Waiting for inactivity to perform halt [Feb 8 12:31:23] Waiting for inactivity to perform halt... [Feb 8 12:31:23] Executing last minute cleanups [Feb 8 12:31:23] == Destroying musiconhold processes [Feb 8 12:31:23] Asterisk cleanly ending (0). [Feb 8 12:31:23] DEBUG[12420]: asterisk.c:1192 quit_handler: Asterisk ending (0).