<--- SIP read from UDP:172.30.0.188:5060 ---> INVITE sip:*800@burnabyasterisk.customer.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKa13d06869C6C3885 From: "4149" ;tag=98C00654-62489A9B To: CSeq: 1 INVITE Call-ID: 4c407148-1604cabf-61cd631a@172.30.0.188 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 294 v=0 o=- 1314381196 1314381196 IN IP4 172.30.0.188 s=Polycom IP Phone c=IN IP4 172.30.0.188 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 <-------------> [2011-08-26 10:53:21] --- (15 headers 13 lines) --- [2011-08-26 10:53:21] == Using UDPTL TOS bits 184 [2011-08-26 10:53:21] == Using UDPTL CoS mark 5 [2011-08-26 10:53:21] Sending to 172.30.0.188:5060 (no NAT) [2011-08-26 10:53:21] Using INVITE request as basis request - 4c407148-1604cabf-61cd631a@172.30.0.188 [2011-08-26 10:53:21] Found peer '4149' for '4149' from 172.30.0.188:5060 [2011-08-26 10:53:21] <--- Reliably Transmitting (no NAT) to 172.30.0.188:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKa13d06869C6C3885;received=172.30.0.188 From: "4149" ;tag=98C00654-62489A9B To: ;tag=as180dca98 Call-ID: 4c407148-1604cabf-61cd631a@172.30.0.188 CSeq: 1 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="665dbfdd" Content-Length: 0 <------------> [2011-08-26 10:53:21] Scheduling destruction of SIP dialog '4c407148-1604cabf-61cd631a@172.30.0.188' in 6400 ms (Method: INVITE) [2011-08-26 10:53:21] <--- SIP read from UDP:172.30.0.188:5060 ---> ACK sip:*800@burnabyasterisk.customer.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKa13d06869C6C3885 From: "4149" ;tag=98C00654-62489A9B To: ;tag=as180dca98 CSeq: 1 ACK Call-ID: 4c407148-1604cabf-61cd631a@172.30.0.188 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Max-Forwards: 70 Content-Length: 0 <-------------> [2011-08-26 10:53:21] --- (12 headers 0 lines) --- [2011-08-26 10:53:21] <--- SIP read from UDP:172.30.0.188:5060 ---> INVITE sip:*800@burnabyasterisk.customer.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK2c849bc91A21C7C From: "4149" ;tag=98C00654-62489A9B To: CSeq: 2 INVITE Call-ID: 4c407148-1604cabf-61cd631a@172.30.0.188 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="4149", realm="asterisk", nonce="665dbfdd", uri="sip:*800@burnabyasterisk.customer.com:5060;user=phone", response="5aa64adf22f2c246e1ccb8ef2c722b25", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 294 v=0 o=- 1314381196 1314381196 IN IP4 172.30.0.188 s=Polycom IP Phone c=IN IP4 172.30.0.188 t=0 0 a=sendrecv m=audio 2222 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 <-------------> [2011-08-26 10:53:21] --- (16 headers 13 lines) --- [2011-08-26 10:53:21] Sending to 172.30.0.188:5060 (no NAT) [2011-08-26 10:53:21] Using INVITE request as basis request - 4c407148-1604cabf-61cd631a@172.30.0.188 [2011-08-26 10:53:21] Found peer '4149' for '4149' from 172.30.0.188:5060 [2011-08-26 10:53:21] == Using SIP RTP TOS bits 184 [2011-08-26 10:53:21] == Using SIP RTP CoS mark 5 [2011-08-26 10:53:21] Found RTP audio format 9 [2011-08-26 10:53:21] Found RTP audio format 0 [2011-08-26 10:53:21] Found RTP audio format 8 [2011-08-26 10:53:21] Found RTP audio format 18 [2011-08-26 10:53:21] Found RTP audio format 101 [2011-08-26 10:53:21] Found audio description format G722 for ID 9 [2011-08-26 10:53:21] Found audio description format PCMU for ID 0 [2011-08-26 10:53:21] Found audio description format PCMA for ID 8 [2011-08-26 10:53:21] Found audio description format G729 for ID 18 [2011-08-26 10:53:21] Found audio description format telephone-event for ID 101 [2011-08-26 10:53:21] Capabilities: us - 0x1004 (ulaw|g722), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1004 (ulaw|g722) [2011-08-26 10:53:21] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2011-08-26 10:53:21] Peer audio RTP is at port 172.30.0.188:2222 [2011-08-26 10:53:21] Looking for *800 in default-super (domain burnabyasterisk.customer.com:5060) [2011-08-26 10:53:21] list_route: hop: [2011-08-26 10:53:21] <--- Transmitting (no NAT) to 172.30.0.188:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK2c849bc91A21C7C;received=172.30.0.188 From: "4149" ;tag=98C00654-62489A9B To: Call-ID: 4c407148-1604cabf-61cd631a@172.30.0.188 CSeq: 2 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> [2011-08-26 10:53:21] -- Executing [*800@default-super:1] Set("SIP/4149-00000049", "GROUP(OUTGOING)=4149") in new stack [2011-08-26 10:53:21] -- Executing [*800@default-super:2] Answer("SIP/4149-00000049", "") in new stack [2011-08-26 10:53:21] Audio is at 5060 [2011-08-26 10:53:21] Adding codec 0x1000 (g722) to SDP [2011-08-26 10:53:21] Adding codec 0x4 (ulaw) to SDP [2011-08-26 10:53:21] Adding non-codec 0x1 (telephone-event) to SDP [2011-08-26 10:53:21] <--- Reliably Transmitting (no NAT) to 172.30.0.188:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bK2c849bc91A21C7C;received=172.30.0.188 From: "4149" ;tag=98C00654-62489A9B To: ;tag=as65f10c6f Call-ID: 4c407148-1604cabf-61cd631a@172.30.0.188 CSeq: 2 INVITE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 258 v=0 o=root 1910856671 1910856671 IN IP4 172.30.0.10 s=Asterisk PBX 1.8.6.0 c=IN IP4 172.30.0.10 t=0 0 m=audio 11562 RTP/AVP 9 0 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [2011-08-26 10:53:21] <--- SIP read from UDP:172.30.0.188:5060 ---> ACK sip:*800@172.30.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.188:5060;branch=z9hG4bKa133a84dB4EF5BF0 From: "4149" ;tag=98C00654-62489A9B To: ;tag=as65f10c6f CSeq: 2 ACK Call-ID: 4c407148-1604cabf-61cd631a@172.30.0.188 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Max-Forwards: 70 Content-Length: 0 <-------------> [2011-08-26 10:53:21] --- (12 headers 0 lines) --- [2011-08-26 10:53:21] -- Executing [*800@default-super:3] Set("SIP/4149-00000049", "CHANNEL(musicclass)=default_default") in new stack [2011-08-26 10:53:21] -- Executing [*800@default-super:4] MixMonitor("SIP/4149-00000049", "conf-1-1314381201.165.WAV,a") in new stack [2011-08-26 10:53:21] -- Executing [*800@default-super:5] MeetMe("SIP/4149-00000049", "1,TMi") in new stack [2011-08-26 10:53:21] -- Playing 'conf-getpin.g722' (language 'en') [2011-08-26 10:53:21] == Begin MixMonitor Recording SIP/4149-00000049 [2011-08-26 10:53:25] -- Recording [2011-08-26 10:53:25] -- Playing 'vm-rec-name.g722' (language 'en') [2011-08-26 10:53:26] <--- SIP read from UDP:172.30.0.189:5060 ---> SUBSCRIBE sip:4109@burnabyasterisk.customer.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK4bc0ef335DCC4722 From: "Boardroom" ;tag=79C4D18D-47ACF9EC To: CSeq: 1 SUBSCRIBE Call-ID: 97b9a179-8d5397f8-d25c2e5f@172.30.0.189 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: message/sipfrag Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [2011-08-26 10:53:26] --- (15 headers 0 lines) --- [2011-08-26 10:53:26] Creating new subscription [2011-08-26 10:53:26] Sending to 172.30.0.189:5060 (no NAT) [2011-08-26 10:53:26] list_route: hop: [2011-08-26 10:53:26] Found peer '4109' for '4109' from 172.30.0.189:5060 [2011-08-26 10:53:26] <--- Transmitting (no NAT) to 172.30.0.189:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK4bc0ef335DCC4722;received=172.30.0.189 From: "Boardroom" ;tag=79C4D18D-47ACF9EC To: ;tag=as44a7aa7a Call-ID: 97b9a179-8d5397f8-d25c2e5f@172.30.0.189 CSeq: 1 SUBSCRIBE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7940bd82" Content-Length: 0 <------------> [2011-08-26 10:53:26] Scheduling destruction of SIP dialog '97b9a179-8d5397f8-d25c2e5f@172.30.0.189' in 6400 ms (Method: SUBSCRIBE) [2011-08-26 10:53:26] <--- SIP read from UDP:172.30.0.189:5060 ---> SUBSCRIBE sip:4109@burnabyasterisk.customer.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bKd16c26e36007BE5 From: "Boardroom" ;tag=79C4D18D-47ACF9EC To: CSeq: 2 SUBSCRIBE Call-ID: 97b9a179-8d5397f8-d25c2e5f@172.30.0.189 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: message/sipfrag Authorization: Digest username="4109", realm="asterisk", nonce="7940bd82", uri="sip:4109@burnabyasterisk.customer.com:5060", response="7ef6b579863bb489e49aa06d9e86572b", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [2011-08-26 10:53:26] --- (16 headers 0 lines) --- [2011-08-26 10:53:26] Creating new subscription [2011-08-26 10:53:26] Sending to 172.30.0.189:5060 (no NAT) [2011-08-26 10:53:26] Found peer '4109' for '4109' from 172.30.0.189:5060 [2011-08-26 10:53:26] Looking for 4109 in default-local (domain burnabyasterisk.customer.com:5060) [2011-08-26 10:53:26] <--- Transmitting (no NAT) to 172.30.0.189:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bKd16c26e36007BE5;received=172.30.0.189 From: "Boardroom" ;tag=79C4D18D-47ACF9EC To: ;tag=as44a7aa7a Call-ID: 97b9a179-8d5397f8-d25c2e5f@172.30.0.189 CSeq: 2 SUBSCRIBE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-08-26 10:53:26] Really destroying SIP dialog '97b9a179-8d5397f8-d25c2e5f@172.30.0.189' Method: SUBSCRIBE [2011-08-26 10:53:27] Reliably Transmitting (no NAT) to 172.30.0.189:5060: OPTIONS sip:4109@172.30.0.189:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a837cda Max-Forwards: 70 From: "asterisk" ;tag=as29115d28 To: Contact: Call-ID: 56d77e9a5dce299b280b4e2e645ec452@172.30.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Date: Fri, 26 Aug 2011 17:53:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-08-26 10:53:27] <--- SIP read from UDP:172.30.0.189:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5a837cda From: "asterisk" ;tag=as29115d28 To: "Boardroom" ;tag=BAA44437-32C58486 CSeq: 102 OPTIONS Call-ID: 56d77e9a5dce299b280b4e2e645ec452@172.30.0.10:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces,100rel,timer,replaces,norefersub User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Content-Length: 0 <-------------> [2011-08-26 10:53:27] --- (14 headers 0 lines) --- [2011-08-26 10:53:27] Really destroying SIP dialog '56d77e9a5dce299b280b4e2e645ec452@172.30.0.10:5060' Method: OPTIONS [2011-08-26 10:53:27] Reliably Transmitting (no NAT) to 172.30.0.195:5060: OPTIONS sip:4145@172.30.0.195:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45bba512 Max-Forwards: 70 From: "asterisk" ;tag=as0385c6e9 To: Contact: Call-ID: 2ca360a8463f456267de84ea082d2d69@172.30.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Date: Fri, 26 Aug 2011 17:53:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-08-26 10:53:27] Reliably Transmitting (no NAT) to 172.30.0.191:5060: OPTIONS sip:4119@172.30.0.191:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK465b4d27 Max-Forwards: 70 From: "asterisk" ;tag=as4dea7524 To: Contact: Call-ID: 5010478e15bda8075eb878255561827c@172.30.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Date: Fri, 26 Aug 2011 17:53:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-08-26 10:53:27] <--- SIP read from UDP:172.30.0.195:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK45bba512 From: "asterisk" ;tag=as0385c6e9 To: "LORA MANOR" ;tag=F32D3C7-DD70894A CSeq: 102 OPTIONS Call-ID: 2ca360a8463f456267de84ea082d2d69@172.30.0.10:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces,100rel,timer,replaces,norefersub User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Content-Length: 0 <-------------> [2011-08-26 10:53:27] --- (14 headers 0 lines) --- [2011-08-26 10:53:27] Reliably Transmitting (no NAT) to 172.30.0.186:5060: OPTIONS sip:4169@172.30.0.186:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK61332eaf Max-Forwards: 70 From: "asterisk" ;tag=as5a2dedaa To: Contact: Call-ID: 21bb161126bbdf314cbb23745d40126a@172.30.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Date: Fri, 26 Aug 2011 17:53:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-08-26 10:53:27] Really destroying SIP dialog '2ca360a8463f456267de84ea082d2d69@172.30.0.10:5060' Method: OPTIONS [2011-08-26 10:53:27] <--- SIP read from UDP:172.30.0.191:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK465b4d27 From: "asterisk" ;tag=as4dea7524 To: "empty station" ;tag=50954409-30E2EC48 CSeq: 102 OPTIONS Call-ID: 5010478e15bda8075eb878255561827c@172.30.0.10:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces,100rel,timer,replaces,norefersub User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Content-Length: 0 <-------------> [2011-08-26 10:53:27] --- (14 headers 0 lines) --- [2011-08-26 10:53:27] Really destroying SIP dialog '5010478e15bda8075eb878255561827c@172.30.0.10:5060' Method: OPTIONS [2011-08-26 10:53:27] <--- SIP read from UDP:172.30.0.186:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK61332eaf From: "asterisk" ;tag=as5a2dedaa To: "4169" ;tag=474B86D0-3AAD32D3 CSeq: 102 OPTIONS Call-ID: 21bb161126bbdf314cbb23745d40126a@172.30.0.10:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces,100rel,timer,replaces,norefersub User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Content-Length: 0 <-------------> [2011-08-26 10:53:27] --- (14 headers 0 lines) --- [2011-08-26 10:53:27] Really destroying SIP dialog '21bb161126bbdf314cbb23745d40126a@172.30.0.10:5060' Method: OPTIONS [2011-08-26 10:53:29] -- Playing 'beep.g722' (language 'en') [2011-08-26 10:53:29] -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-1-7 format: sln, 0x8dd1308 [2011-08-26 10:53:30] Reliably Transmitting (no NAT) to 172.30.0.187:5060: OPTIONS sip:4168@172.30.0.187:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5b3c096d Max-Forwards: 70 From: "asterisk" ;tag=as0e72eb5b To: Contact: Call-ID: 3cf751d25dc50ca23f78bff84c162822@172.30.0.10:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX (asterisk) Date: Fri, 26 Aug 2011 17:53:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2011-08-26 10:53:30] <--- SIP read from UDP:172.30.0.187:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.30.0.10:5060;branch=z9hG4bK5b3c096d From: "asterisk" ;tag=as0e72eb5b To: "4168" ;tag=51B7A95C-EE394061 CSeq: 102 OPTIONS Call-ID: 3cf751d25dc50ca23f78bff84c162822@172.30.0.10:5060 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces,100rel,timer,replaces,norefersub User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml Accept-Encoding: identity Content-Length: 0 <-------------> [2011-08-26 10:53:30] --- (14 headers 0 lines) --- [2011-08-26 10:53:30] Really destroying SIP dialog '3cf751d25dc50ca23f78bff84c162822@172.30.0.10:5060' Method: OPTIONS [2011-08-26 10:53:31] -- User ended message by pressing # [2011-08-26 10:53:31] -- Playing 'auth-thankyou.g722' (language 'en') [2011-08-26 10:53:31] -- Playing 'vm-review.g722' (language 'en') [2011-08-26 10:53:33] <--- SIP read from UDP:172.30.0.189:5060 ---> SUBSCRIBE sip:4109@burnabyasterisk.customer.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bKf7a5103dDB10591C From: "Boardroom" ;tag=A8E61C3A-700710D1 To: CSeq: 1 SUBSCRIBE Call-ID: efb07ae3-bffa2b52-83dc2a29@172.30.0.189 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: message/sipfrag Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [2011-08-26 10:53:33] --- (15 headers 0 lines) --- [2011-08-26 10:53:33] Creating new subscription [2011-08-26 10:53:33] Sending to 172.30.0.189:5060 (no NAT) [2011-08-26 10:53:33] list_route: hop: [2011-08-26 10:53:33] Found peer '4109' for '4109' from 172.30.0.189:5060 [2011-08-26 10:53:33] <--- Transmitting (no NAT) to 172.30.0.189:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bKf7a5103dDB10591C;received=172.30.0.189 From: "Boardroom" ;tag=A8E61C3A-700710D1 To: ;tag=as549b8acd Call-ID: efb07ae3-bffa2b52-83dc2a29@172.30.0.189 CSeq: 1 SUBSCRIBE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7af6145e" Content-Length: 0 <------------> [2011-08-26 10:53:33] Scheduling destruction of SIP dialog 'efb07ae3-bffa2b52-83dc2a29@172.30.0.189' in 6400 ms (Method: SUBSCRIBE) [2011-08-26 10:53:33] <--- SIP read from UDP:172.30.0.189:5060 ---> SUBSCRIBE sip:4109@burnabyasterisk.customer.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK7d083128EE52640F From: "Boardroom" ;tag=A8E61C3A-700710D1 To: CSeq: 2 SUBSCRIBE Call-ID: efb07ae3-bffa2b52-83dc2a29@172.30.0.189 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: missed-call-summary User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.5.0508 Accept-Language: en-us,en;q=0.9 Accept: message/sipfrag Authorization: Digest username="4109", realm="asterisk", nonce="7af6145e", uri="sip:4109@burnabyasterisk.customer.com:5060", response="4418824353d4417741db19c38b23e419", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 <-------------> [2011-08-26 10:53:33] --- (16 headers 0 lines) --- [2011-08-26 10:53:33] Creating new subscription [2011-08-26 10:53:33] Sending to 172.30.0.189:5060 (no NAT) [2011-08-26 10:53:33] Found peer '4109' for '4109' from 172.30.0.189:5060 [2011-08-26 10:53:33] Looking for 4109 in default-local (domain burnabyasterisk.customer.com:5060) [2011-08-26 10:53:33] <--- Transmitting (no NAT) to 172.30.0.189:5060 ---> SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 172.30.0.189:5060;branch=z9hG4bK7d083128EE52640F;received=172.30.0.189 From: "Boardroom" ;tag=A8E61C3A-700710D1 To: ;tag=as549b8acd Call-ID: efb07ae3-bffa2b52-83dc2a29@172.30.0.189 CSeq: 2 SUBSCRIBE Server: Asterisk PBX (asterisk) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> [2011-08-26 10:53:33] Really destroying SIP dialog 'efb07ae3-bffa2b52-83dc2a29@172.30.0.189' Method: SUBSCRIBE burnabyasterisk*CLI> Disconnected from Asterisk server [Aug 26 10:53:35] Executing last minute cleanups Asterisk ending (0).