; the console log of a failed sip transfer ; 1. The caller is "A CUSTOMER" <9058431234> ; 2. The callee is Agent/329 ; 3. The phone SIP/329 was used to answer the call. ; 4. A warm transfer to SIP/328 was attempted. -- Accepting AUTHENTICATED call from 216.7.201.43, requested format = 4, actual format = 4 -- Executing NoOp("IAX2/oce01pbx@216.7.201.43:4569/10", "20051212-094556 in-oce call for itd from "A CUSTOMER" <9058431234> for s") in new stack -- Executing Goto("IAX2/oce01pbx@216.7.201.43:4569/10", "itd01-main|s|1") in new stack -- Goto (itd01-main,s,1) -- Executing Answer("IAX2/oce01pbx@216.7.201.43:4569/10", "") in new stack -- Executing Wait("IAX2/oce01pbx@216.7.201.43:4569/10", "1") in new stack 11 headers, 0 lines Destroying call '6bf9980c034ea11b03727ce85ce00aab@142.46.202.202' -- Executing ResponseTimeout("IAX2/oce01pbx@216.7.201.43:4569/10", "45") in new stack -- Set Response Timeout to 45 -- Executing SetMusicOnHold("IAX2/oce01pbx@216.7.201.43:4569/10", "default") in new stack -- Executing BackGround("IAX2/oce01pbx@216.7.201.43:4569/10", "itd/welcome") in new stack -- Playing 'itd/welcome' (language 'en') 11 headers, 0 lines Destroying call '1b88d5ac2738cfc3705e835a0a6e8dcc@142.46.202.202' 11 headers, 0 lines Sip read: 0 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6f245db7 From: "asterisk" ;tag=as002ca35a To: Contact: Call-ID: 119ecdec4fcf1e8322c6c69f2d27a398@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.213:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6f245db7 From: "asterisk" ;tag=as002ca35a To: ;tag=D81DE714-EDD7B799 CSeq: 102 OPTIONS Call-ID: 119ecdec4fcf1e8322c6c69f2d27a398@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '119ecdec4fcf1e8322c6c69f2d27a398@142.46.202.202' == CDR updated on IAX2/oce01pbx@216.7.201.43:4569/10 -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/10", "silence/1") in new stack -- Playing 'silence/1' (language 'en') -- Executing SetVar("IAX2/oce01pbx@216.7.201.43:4569/10", "QUEUESOURCE=itd01-main") in new stack -- Executing SetVar("IAX2/oce01pbx@216.7.201.43:4569/10", "CIDPREFIX=HelpDesk") in new stack -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/10", "itd/this-call-may-be-recorded-for-quality-purposes") in new stack -- Playing 'itd/this-call-may-be-recorded-for-quality-purposes' (language 'en') 11 headers, 0 lines Destroying call '03842f7b4cd784c31145fdb54345ec5d@142.46.202.202' Destroying call '27cc5b96406642f8204cf2a065aa4a99@142.46.202.202' 11 headers, 0 lines Destroying call '6ccb8bb80d5f5f4855b5dc0f5aabae0b@142.46.202.202' -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/10", "silence/1") in new stack -- Playing 'silence/1' (language 'en') 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:147.135.8.128 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK17908145 From: "asterisk" ;tag=as517b027b To: Contact: Call-ID: 42f47db42894c7a27e9409de01b4e1ed@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 147.135.8.128:5060 Sip read: SIP/2.0 200 OK Call-ID: 42f47db42894c7a27e9409de01b4e1ed@142.46.202.202 CSeq: 102 OPTIONS From: "asterisk" ;tag=as517b027b To: Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK17908145 Supported: 100rel Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK Accept: application/sdp Accept-Encoding: Accept-Language: en Content-Length: 0 12 headers, 0 lines Destroying call '42f47db42894c7a27e9409de01b4e1ed@142.46.202.202' -- Executing Goto("IAX2/oce01pbx@216.7.201.43:4569/10", "itd01-queue-helpdesk|s|1") in new stack -- Goto (itd01-queue-helpdesk,s,1) -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/10", "silence/1") in new stack -- Playing 'silence/1' (language 'en') -- Executing ResponseTimeout("IAX2/oce01pbx@216.7.201.43:4569/10", "5") in new stack -- Set Response Timeout to 5 -- Executing BackGround("IAX2/oce01pbx@216.7.201.43:4569/10", "itd/itd-helpdesk-get_ticket") in new stack -- Playing 'itd/itd-helpdesk-get_ticket' (language 'en') Sip read: 0 headers, 0 lines 12 headers, 0 lines Destroying call '3e1558785d0c145a4aac97eb594f1752@142.46.202.202' == CDR updated on IAX2/oce01pbx@216.7.201.43:4569/10 -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/10", "silence/1") in new stack -- Playing 'silence/1' (language 'en') 11 headers, 0 lines Destroying call '2161a53a57fa84a04cddcb7a6a2a51f5@142.46.202.202' -- Executing SetVar("IAX2/oce01pbx@216.7.201.43:4569/10", "CIDPREFIX=HD/24443") in new stack -- Executing BackGround("IAX2/oce01pbx@216.7.201.43:4569/10", "auth-thankyou") in new stack -- Playing 'auth-thankyou' (language 'en') 11 headers, 0 lines Destroying call '50176b823aceab5e07f321490c780345@142.46.202.202' -- Executing BackGround("IAX2/oce01pbx@216.7.201.43:4569/10", "itd/itd-helpdesk-got_ticket") in new stack -- Playing 'itd/itd-helpdesk-got_ticket' (language 'en') 11 headers, 0 lines Destroying call '603c8d890e0fefe7607f9f693837122a@142.46.202.202' Sip read: 0 headers, 0 lines -- Executing Goto("IAX2/oce01pbx@216.7.201.43:4569/10", "joinqueue|1") in new stack -- Goto (itd01-queue-helpdesk,joinqueue,1) -- Executing SetCIDName("IAX2/oce01pbx@216.7.201.43:4569/10", "HD/24443/A CUSTOMER") in new stack -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/10", "itd/itd-helpdesk-hold_menu") in new stack -- Playing 'itd/itd-helpdesk-hold_menu' (language 'en') 11 headers, 0 lines Destroying call '0bc916e3640fe348742658c84697c241@142.46.202.202' 12 headers, 0 lines Reliably Transmitting: Destroying call '27cc5b96406642f8204cf2a065aa4a99@142.46.202.202' Sip read: 0 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: Destroying call '1864cea57534092a180696b611ab57fc@142.46.202.202' 11 headers, 0 lines ;Agent/329 on SIP/329 answers the call Destroying call '49a9ee0124df51947f4174bd695f6c98@142.46.202.202' -- Executing Queue("IAX2/oce01pbx@216.7.201.43:4569/10", "itd-helpdesk") in new stack -- Started music on hold, class 'default', on IAX2/oce01pbx@216.7.201.43:4569/10 -- outgoing agentcall, to agent '329', on 'Local/329@itd01-internal-d9a8,1' -- Executing Macro("Local/329@itd01-internal-d9a8,2", "multi-dial|SIP/329|SIP/3291|Zap/g1/6138682251") in new stack -- Executing NoOp("Local/329@itd01-internal-d9a8,2", "Incoming CID: "HD/24443/A CUSTOMER" <9058431234>") in new stack -- Executing GotoIf("Local/329@itd01-internal-d9a8,2", "0?3:20") in new stack -- Goto (macro-multi-dial,s,20) -- Executing SetVar("Local/329@itd01-internal-d9a8,2", "DIALSTR=SIP/329&SIP/3291&Zap/g1/6138682251") in new stack -- Executing Goto("Local/329@itd01-internal-d9a8,2", "s|50") in new stack -- Goto (macro-multi-dial,s,50) -- Executing Dial("Local/329@itd01-internal-d9a8,2", "SIP/329&SIP/3291&Zap/g1/6138682251|30|r") in new stack We're at 142.46.202.202 port 10034 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting: INVITE sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7e212556 From: "HD/24443/A CUSTOMER" ;tag=as68cc696f To: Contact: Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8840 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.213:5060 -- Called 329 -- Called Agent/329 Dec 12 09:46:30 WARNING[8840]: chan_sip.c:1401 create_addr: No such host: 3291 Destroying call '70edfbbb31cde1b82e86694f152ef142@142.46.202.202' Dec 12 09:46:30 NOTICE[8840]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP' -- Called g1/6138682251 -- Agent/329 is ringing Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7e212556 From: "HD/24443/A CUSTOMER" ;tag=as68cc696f To: ;tag=F2B22CDD-B293F366 CSeq: 102 INVITE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 9 headers, 0 lines Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7e212556 From: "HD/24443/A CUSTOMER" ;tag=as68cc696f To: ;tag=F2B22CDD-B293F366 CSeq: 102 INVITE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Allow-Events: talk,hold,conference Content-Length: 0 10 headers, 0 lines -- SIP/329-43d4 is ringing -- Agent/329 is ringing Sip read: 0 headers, 0 lines Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7e212556 From: "HD/24443/A CUSTOMER" ;tag=as68cc696f To: ;tag=F2B22CDD-B293F366 CSeq: 102 INVITE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1134398781 1134398781 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 10.1.0.213 t=0 0 m=audio 2232 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 11 headers, 8 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.213:2232 Found description format PCMU Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.213, port 5060 Transmitting: ACK sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3f8fdc80 From: "HD/24443/A CUSTOMER" ;tag=as68cc696f To: ;tag=F2B22CDD-B293F366 Contact: Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.1.0.213:5060 -- SIP/329-43d4 answered Local/329@itd01-internal-d9a8,2 -- Hungup 'Zap/5-1' -- Agent/329 answered IAX2/oce01pbx@216.7.201.43:4569/10 -- Stopped music on hold on IAX2/oce01pbx@216.7.201.43:4569/10 11 headers, 0 lines Destroying call '6c705b601bed8c4b5763101901615e98@142.46.202.202' -- Attempting native bridge of Zap/2-1 and Zap/3-1 12 headers, 0 lines Destroying call '27cc5b96406642f8204cf2a065aa4a99@142.46.202.202' 12 headers, 0 lines Destroying call '3e1558785d0c145a4aac97eb594f1752@142.46.202.202' 11 headers, 0 lines Destroying call '44fbce1b75dc165733eda363316c297b@142.46.202.202' Sip read: 0 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:328@10.1.0.205 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1306552b From: "asterisk" ;tag=as30fb3e63 To: Contact: Call-ID: 16e656ac247f3e5e25c590970ff10012@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.205:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1306552b From: "asterisk" ;tag=as30fb3e63 To: ;tag=2837E22C-6AAE0A5 CSeq: 102 OPTIONS Call-ID: 16e656ac247f3e5e25c590970ff10012@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '16e656ac247f3e5e25c590970ff10012@142.46.202.202' Sip read: INVITE sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKd2c6eb08FA2FA7D7 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 1 INVITE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 180 v=0 o=- 1134398782 1134398782 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2232 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 14 headers, 8 lines Using latest request as basis request Sending to 10.1.0.213 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:2232 Found description format PCMU Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 142.46.202.202 port 10034 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKd2c6eb08FA2FA7D7 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8841 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKda4f767a57608A61 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 1 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines 11 headers, 0 lines == Manager 'asttapi' logged on from 142.46.202.202 -- Remote UNIX connection == Manager 'localscripts' logged on from 142.46.202.202 == Manager 'localscripts' logged off from 142.46.202.202 == Manager 'asttapi' logged off from 142.46.202.202 -- Remote UNIX connection disconnected Sip read: 0 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7ce8e68f From: "asterisk" ;tag=as5949deb7 To: Contact: Call-ID: 332b63b40b642f1e1dcbbd3012d18894@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.213:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7ce8e68f From: "asterisk" ;tag=as5949deb7 To: ;tag=CE9CECE-450736FB CSeq: 102 OPTIONS Call-ID: 332b63b40b642f1e1dcbbd3012d18894@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '332b63b40b642f1e1dcbbd3012d18894@142.46.202.202' 12 headers, 0 lines Sip read: 0 headers, 0 lines 11 headers, 0 lines Sip read: 0 headers, 0 lines 11 headers, 0 lines Sip read: INVITE sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK8a0e9425839D30B0 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 2 INVITE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 233 v=0 o=- 1134398783 1134398783 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 10.1.0.213 t=0 0 m=audio 2232 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 14 headers, 10 lines Using latest request as basis request Sending to 10.1.0.213 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.213:2232 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 142.46.202.202 port 10034 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK8a0e9425839D30B0 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8842 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKac33ac5f742E9062 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 2 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines 11 headers, 0 lines Sip read: 0 headers, 0 lines Sip read: 0 headers, 0 lines Sip read: INVITE sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc0b84e293C69A64 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 3 INVITE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 180 v=0 o=- 1134398784 1134398784 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2232 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 14 headers, 8 lines Using latest request as basis request Sending to 10.1.0.213 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:2232 Found description format PCMU Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 142.46.202.202 port 10034 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc0b84e293C69A64 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8843 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: INVITE sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc0b84e293C69A64 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 3 INVITE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 180 v=0 o=- 1134398784 1134398784 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2232 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 14 headers, 8 lines Ignoring this request We're at 142.46.202.202 port 10034 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc0b84e293C69A64 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8844 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK63c6ac03AFC68F36 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 3 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines Retransmitting #1 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc0b84e293C69A64 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8843 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK6b6dfc6d45C86358 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 3 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines Retransmitting #2 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc0b84e293C69A64 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8843 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK316019e7B94BDF4A From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 3 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines Destroying call 'dadc9d35-b7b3b67f-5c1b7b2e@192.168.1.50' Retransmitting #3 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc0b84e293C69A64 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8843 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK5ed822f1E58BF8C From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 3 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines Retransmitting #4 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc0b84e293C69A64 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8843 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Dec 12 09:47:58 NOTICE[12125]: chan_sip.c:4017 sip_reregister: -- Re-registration for 19024826475@sphone.vopr.vonage.net -- parse_srv: SRV mapped to host sphone.vopr.vonage.net, port 5061 12 headers, 0 lines Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK10521a0bAD9DD49E From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 3 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc0b84e293C69A64 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8843 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK7afb85b5B5D52300 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 3 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines 11 headers, 0 lines Dec 12 09:48:00 WARNING[12125]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 for seqno 3 (Non-critical Response) 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:323@10.1.0.217 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK36cb6148 From: "asterisk" ;tag=as589e446b To: Contact: Call-ID: 7b93ebe33fdce7fe0247055421dd6d31@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.217:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK36cb6148 From: "asterisk" ;tag=as589e446b To: ;tag=67B8085E-C271EA25 CSeq: 102 OPTIONS Call-ID: 7b93ebe33fdce7fe0247055421dd6d31@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '7b93ebe33fdce7fe0247055421dd6d31@142.46.202.202' Sip read: 0 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK09645c41 From: "asterisk" ;tag=as6e78e826 To: Contact: Call-ID: 2f34367a4eb76a2666824fa62f26c399@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.213:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK09645c41 From: "asterisk" ;tag=as6e78e826 To: ;tag=D23D28B9-F750332 CSeq: 102 OPTIONS Call-ID: 2f34367a4eb76a2666824fa62f26c399@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '2f34367a4eb76a2666824fa62f26c399@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3060@64.26.161.135:10254 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6e705238 From: "asterisk" ;tag=as505e7fd6 To: Contact: Call-ID: 166066b95d473bef2655bc9c11f4529d@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 64.26.161.135:10254 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6e705238 From: "asterisk" ;tag=as505e7fd6 To: ;tag=F4521E15-8C58FB4A Call-ID: 166066b95d473bef2655bc9c11f4529d@142.46.202.202 CSeq: 102 OPTIONS Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Content-Length: 0 10 headers, 0 lines Destroying call '166066b95d473bef2655bc9c11f4529d@142.46.202.202' Destroying call 'a214b801-f76d472f-b8b357ea@10.1.0.208' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:330@192.168.1.104 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6fcf97fa From: "asterisk" ;tag=as3afa0135 To: Contact: Call-ID: 5d4a8e2f35e553f542b4267f0c7ec4a9@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 72.56.142.31:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6fcf97fa From: "asterisk" ;tag=as3afa0135 To: ;tag=E9CF73C7-FEF9B80E CSeq: 102 OPTIONS Call-ID: 5d4a8e2f35e553f542b4267f0c7ec4a9@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '5d4a8e2f35e553f542b4267f0c7ec4a9@142.46.202.202' Sip read: INVITE sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK7af641b4F49BA113 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 4 INVITE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 233 v=0 o=- 1134398785 1134398785 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 10.1.0.213 t=0 0 m=audio 2232 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 14 headers, 10 lines Using latest request as basis request Sending to 10.1.0.213 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.213:2232 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 142.46.202.202 port 10034 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK7af641b4F49BA113 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 4 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8845 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK57373f0694A64FFD From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 4 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:147.135.8.128 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK36ff6f59 From: "asterisk" ;tag=as3dd3013a To: Contact: Call-ID: 3da42e62372e45512597a4ad5404951b@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 147.135.8.128:5060 Sip read: SIP/2.0 200 OK Call-ID: 3da42e62372e45512597a4ad5404951b@142.46.202.202 CSeq: 102 OPTIONS From: "asterisk" ;tag=as3dd3013a To: Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK36ff6f59 Supported: 100rel Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK Accept: application/sdp Accept-Encoding: Accept-Language: en Content-Length: 0 12 headers, 0 lines Destroying call 'cea8d355-53b27cfb-8a51ac44@10.1.0.203' Destroying call '3da42e62372e45512597a4ad5404951b@142.46.202.202' == Manager 'localscripts' logged on from 142.46.202.202 == Manager 'localscripts' logged off from 142.46.202.202 Sip read: 0 headers, 0 lines Dec 12 09:48:14 NOTICE[12125]: chan_sip.c:4017 sip_reregister: -- Re-registration for 19024826475@sphone.vopr.vonage.net -- parse_srv: SRV mapped to host sphone.vopr.vonage.net, port 5061 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sphone.vopr.vonage.net SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK25f325c8 From: ;tag=as19caa786 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14785 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="19024826475", realm="216.115.25.198", algorithm=MD5, uri="sip:216.115.25.198", nonce="1396097638", response="d76fab465571c6be048e9135873d194a", opaque="" Expires: 600 Contact: Event: registration Content-Length: 0 (no NAT) to 216.115.25.198:5061 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK25f325c8 From: ;tag=as19caa786 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14785 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:48:14 NOTICE[12125]: chan_sip.c:6831 handle_response: Outbound Registration: Expiry for sphone.vopr.vonage.net is 20 sec (Scheduling reregistration in 15999 ms) Destroying call '27cc5b96406642f8204cf2a065aa4a99@142.46.202.202' Dec 12 09:48:15 NOTICE[12125]: chan_sip.c:4017 sip_reregister: -- Re-registration for 6109770133@sip.broadvoice.com 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK63345215 From: ;tag=as10dbf802 To: Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9889 REGISTER User-Agent: Asterisk PBX Authorization: Digest username="6109770133", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="1134164694288", response="60b15182fedf77b072cb8ebfa217b4e7", opaque="" Expires: 600 Contact: Event: registration Content-Length: 0 (no NAT) to 147.135.8.128:5060 Sip read: SIP/2.0 200 OK Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9889 REGISTER From: ;tag=as10dbf802 To: Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK63345215 Contact: Expires: 30 Authorization: Digest username="6109770133", realm="BroadWorks", algorithm=MD5, uri="sip:sip.broadvoice.com", nonce="1134164694288", response="60b15182fedf77b072cb8ebfa217b4e7", opaque="" Event: registration User-Agent: Asterisk PBX Content-Length: 0 12 headers, 0 lines Dec 12 09:48:15 NOTICE[12125]: chan_sip.c:6831 handle_response: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23999 ms) Destroying call '3e1558785d0c145a4aac97eb594f1752@142.46.202.202' Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.204;branch=z9hG4bK6b84ab142E8763DB From: "307" ;tag=18A52122-C4427213 To: CSeq: 131 REGISTER Call-ID: aad51016-a64b270c-c2d1844d@10.1.0.204 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Authorization: Digest username="307", realm="asterisk", nonce="1436e2e7", uri="sip:142.46.202.202:5060", response="fa40a97939a6cecc8df69368333cbfec", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 10.1.0.204 : 5060 (NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.204;branch=z9hG4bK6b84ab142E8763DB From: "307" ;tag=18A52122-C4427213 To: Call-ID: aad51016-a64b270c-c2d1844d@10.1.0.204 CSeq: 131 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.204:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.0.204;branch=z9hG4bK6b84ab142E8763DB From: "307" ;tag=18A52122-C4427213 To: ;tag=as65280cef Call-ID: aad51016-a64b270c-c2d1844d@10.1.0.204 CSeq: 131 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="3b4348d9" Content-Length: 0 to 10.1.0.204:5060 Scheduling destruction of call 'aad51016-a64b270c-c2d1844d@10.1.0.204' in 15000 ms Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.204;branch=z9hG4bK5ef622aa5CCBC541 From: "307" ;tag=18A52122-C4427213 To: CSeq: 132 REGISTER Call-ID: aad51016-a64b270c-c2d1844d@10.1.0.204 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Authorization: Digest username="307", realm="asterisk", nonce="3b4348d9", uri="sip:142.46.202.202:5060", response="8e3e5ddeb100021ccc5d3b138ef2d095", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 10.1.0.204 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.204;branch=z9hG4bK5ef622aa5CCBC541 From: "307" ;tag=18A52122-C4427213 To: Call-ID: aad51016-a64b270c-c2d1844d@10.1.0.204 CSeq: 132 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.204:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.204;branch=z9hG4bK5ef622aa5CCBC541 From: "307" ;tag=18A52122-C4427213 To: ;tag=as65280cef Call-ID: aad51016-a64b270c-c2d1844d@10.1.0.204 CSeq: 132 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: ;expires=3600 Date: Mon, 12 Dec 2005 14:48:15 GMT Content-Length: 0 to 10.1.0.204:5060 Scheduling destruction of call 'aad51016-a64b270c-c2d1844d@10.1.0.204' in 15000 ms Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.222;branch=z9hG4bKf430b26B4E4265F From: "306" ;tag=448B4E88-4443BBA7 To: CSeq: 445 REGISTER Call-ID: cabc8274-67f0c32e-f49192dd@10.1.0.222 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Authorization: Digest username="306", realm="asterisk", nonce="3696234b", uri="sip:142.46.202.202:5060", response="c11d6799c3c0e6fc558a1789b95c3909", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 10.1.0.222 : 5060 (NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.222;branch=z9hG4bKf430b26B4E4265F From: "306" ;tag=448B4E88-4443BBA7 To: Call-ID: cabc8274-67f0c32e-f49192dd@10.1.0.222 CSeq: 445 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.222:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.0.222;branch=z9hG4bKf430b26B4E4265F From: "306" ;tag=448B4E88-4443BBA7 To: ;tag=as312c8f3a Call-ID: cabc8274-67f0c32e-f49192dd@10.1.0.222 CSeq: 445 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="76ef099a" Content-Length: 0 to 10.1.0.222:5060 Scheduling destruction of call 'cabc8274-67f0c32e-f49192dd@10.1.0.222' in 15000 ms Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.222;branch=z9hG4bKb740ea00646E1249 From: "306" ;tag=448B4E88-4443BBA7 To: CSeq: 446 REGISTER Call-ID: cabc8274-67f0c32e-f49192dd@10.1.0.222 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Authorization: Digest username="306", realm="asterisk", nonce="76ef099a", uri="sip:142.46.202.202:5060", response="71ef0ab0ba544869e15a475b6c6da841", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 10.1.0.222 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.222;branch=z9hG4bKb740ea00646E1249 From: "306" ;tag=448B4E88-4443BBA7 To: Call-ID: cabc8274-67f0c32e-f49192dd@10.1.0.222 CSeq: 446 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.222:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.222;branch=z9hG4bKb740ea00646E1249 From: "306" ;tag=448B4E88-4443BBA7 To: ;tag=as312c8f3a Call-ID: cabc8274-67f0c32e-f49192dd@10.1.0.222 CSeq: 446 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: ;expires=3600 Date: Mon, 12 Dec 2005 14:48:16 GMT Content-Length: 0 to 10.1.0.222:5060 Scheduling destruction of call 'cabc8274-67f0c32e-f49192dd@10.1.0.222' in 15000 ms 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:326@10.1.0.173 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK29ef4c0e From: "asterisk" ;tag=as7cd055cc To: Contact: Call-ID: 469d356c7fe301ec3cdb7fe81b61de28@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.173:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK29ef4c0e From: "asterisk" ;tag=as7cd055cc To: ;tag=D6AD7F88-93491155 CSeq: 102 OPTIONS Call-ID: 469d356c7fe301ec3cdb7fe81b61de28@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '469d356c7fe301ec3cdb7fe81b61de28@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:310@10.1.0.207 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3f34ee14 From: "asterisk" ;tag=as4db6564a To: Contact: Call-ID: 2843035a41b3434a6fbb18114b3e1183@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.207:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3f34ee14 From: "asterisk" ;tag=as4db6564a To: ;tag=92C76868-D92116BD CSeq: 102 OPTIONS Call-ID: 2843035a41b3434a6fbb18114b3e1183@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Content-Length: 0 10 headers, 0 lines Destroying call '2843035a41b3434a6fbb18114b3e1183@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3330@10.1.4.51 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1823cc28 From: "asterisk" ;tag=as64c57549 To: Contact: Call-ID: 0b6d3fe70fffc60f7f9499e9520a7553@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.4.51:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1823cc28 From: "asterisk" ;tag=as64c57549 To: ;tag=6C3A2C76-A97A7C1B CSeq: 102 OPTIONS Call-ID: 0b6d3fe70fffc60f7f9499e9520a7553@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '0b6d3fe70fffc60f7f9499e9520a7553@142.46.202.202' == Manager 'asttapi' logged on from 142.46.202.202 -- Remote UNIX connection == Manager 'asttapi' logged off from 142.46.202.202 -- Remote UNIX connection disconnected Sip read: 0 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:26912@10.1.0.219 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK25db9c32 From: "asterisk" ;tag=as0705c3ba To: Contact: Call-ID: 15b833b65b2c147523392a3c3dfb3710@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.219:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK25db9c32 From: "asterisk" ;tag=as0705c3ba To: ;tag=D5B6A305-1D000188 CSeq: 102 OPTIONS Call-ID: 15b833b65b2c147523392a3c3dfb3710@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '15b833b65b2c147523392a3c3dfb3710@142.46.202.202' Sip read: INVITE sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc4c40fa8D1D6AFF7 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 5 INVITE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 180 v=0 o=- 1134398786 1134398786 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2232 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 14 headers, 8 lines Using latest request as basis request Sending to 10.1.0.213 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:2232 Found description format PCMU Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 142.46.202.202 port 10034 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKc4c40fa8D1D6AFF7 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 5 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8840 8846 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10034 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 Sip read: ACK sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKe40b9c1a43687F81 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 5 ACK Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines 11 headers, 2 lines ;SIP/329 calls SIP/328 and explains she wants to transfer the call ; (A "warm hand-off") Sip read: INVITE sip:328@142.46.202.202:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK87c9c0dc3839611B From: "329" ;tag=FA854A02-E46D187F To: CSeq: 1 INVITE Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 233 v=0 o=- 1134398893 1134398893 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 10.1.0.213 t=0 0 m=audio 2228 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 14 headers, 10 lines Using latest request as basis request Sending to 10.1.0.213 : 5060 (NAT) Reliably Transmitting (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK87c9c0dc3839611B;received=10.1.0.213;rport=5060 From: "329" ;tag=FA854A02-E46D187F To: ;tag=as79e51313 Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="7920039a" Content-Length: 0 to 10.1.0.213:5060 Scheduling destruction of call 'a22e6e6e-4289c950-f0a17b45@10.1.0.213' in 15000 ms Found user '329' 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:306@10.1.0.222 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6a27fe1c From: "asterisk" ;tag=as05871beb To: Contact: Call-ID: 6030bd45593a86b24ced4d8c7324ff93@142.46.202.202 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/1 (no NAT) to 10.1.0.222:5060 Scheduling destruction of call '6030bd45593a86b24ced4d8c7324ff93@142.46.202.202' in 15000 ms Sip read: ACK sip:328@142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK87c9c0dc3839611B From: "329" ;tag=FA854A02-E46D187F To: ;tag=as79e51313 CSeq: 1 ACK Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines Sip read: INVITE sip:328@142.46.202.202:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK8f5f47494531DD04 From: "329" ;tag=FA854A02-E46D187F To: CSeq: 2 INVITE Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="329", realm="asterisk", nonce="7920039a", uri="sip:328@142.46.202.202:5060;user=phone", response="a2048d96b972d5c6ec4e745fc146e218", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 233 v=0 o=- 1134398893 1134398893 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 10.1.0.213 t=0 0 m=audio 2228 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 15 headers, 10 lines Using latest request as basis request Sending to 10.1.0.213 : 5060 (NAT) Found user '329' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.213:2228 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 328 in itd01-sip list_route: hop: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK8f5f47494531DD04;received=10.1.0.213;rport=5060 From: "329" ;tag=FA854A02-E46D187F To: Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.213:5060 -- Executing NoOp("SIP/329-44f6", "20051212-094827 itd01-sip calling 328 from Nicole Clientcare <329>") in new stack -- Executing Goto("SIP/329-44f6", "itd01-internal|328|1") in new stack -- Goto (itd01-internal,328,1) -- Executing Macro("SIP/329-44f6", "multi-dial|SIP/328|SIP/3281|Zap/g1/6132667741") in new stack -- Executing NoOp("SIP/329-44f6", "Incoming CID: Nicole Clientcare <329>") in new stack -- Executing GotoIf("SIP/329-44f6", "0?3:20") in new stack -- Goto (macro-multi-dial,s,20) -- Executing SetVar("SIP/329-44f6", "DIALSTR=SIP/328&SIP/3281&Zap/g1/6132667741") in new stack -- Executing Goto("SIP/329-44f6", "s|50") in new stack -- Goto (macro-multi-dial,s,50) -- Executing Dial("SIP/329-44f6", "SIP/328&SIP/3281&Zap/g1/6132667741|30|r") in new stack We're at 142.46.202.202 port 10056 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 13 lines Reliably Transmitting: INVITE sip:328@10.1.0.205 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7883cc0a From: "Nicole Clientcare" ;tag=as52811732 To: Contact: Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 8904 8904 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10056 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.205:5060 -- Called 328 Dec 12 09:48:27 WARNING[8904]: chan_sip.c:1401 create_addr: No such host: 3281 Destroying call '3a9644173c671799787be35755d593b9@142.46.202.202' Dec 12 09:48:27 NOTICE[8904]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP' -- Called g1/6132667741 Transmitting (NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK8f5f47494531DD04;received=10.1.0.213;rport=5060 From: "329" ;tag=FA854A02-E46D187F To: ;tag=as6273cf46 Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.213:5060 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7883cc0a From: "Nicole Clientcare" ;tag=as52811732 To: ;tag=E104C494-C731B3ED CSeq: 102 INVITE Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 9 headers, 0 lines Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7883cc0a From: "Nicole Clientcare" ;tag=as52811732 To: ;tag=E104C494-C731B3ED CSeq: 102 INVITE Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Allow-Events: talk,hold,conference Content-Length: 0 10 headers, 0 lines -- SIP/328-e85a is ringing Transmitting (NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK8f5f47494531DD04;received=10.1.0.213;rport=5060 From: "329" ;tag=FA854A02-E46D187F To: ;tag=as6273cf46 Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.213:5060 Sip read: 0 headers, 0 lines Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7883cc0a From: "Nicole Clientcare" ;tag=as52811732 To: ;tag=E104C494-C731B3ED CSeq: 102 INVITE Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1134398868 1134398868 IN IP4 10.1.0.205 s=Polycom IP Phone c=IN IP4 10.1.0.205 t=0 0 m=audio 2230 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 11 headers, 8 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.205:2230 Found description format PCMU Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.205, port 5060 Transmitting: ACK sip:328@10.1.0.205 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK14f6c16c From: "Nicole Clientcare" ;tag=as52811732 To: ;tag=E104C494-C731B3ED Contact: Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.1.0.205:5060 -- SIP/328-e85a answered SIP/329-44f6 -- Hungup 'Zap/2-1' We're at 142.46.202.202 port 10006 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK8f5f47494531DD04;received=10.1.0.213;rport=5060 From: "329" ;tag=FA854A02-E46D187F To: ;tag=as6273cf46 Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=root 8904 8904 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10006 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.1.0.213:5060 -- Attempting native bridge of SIP/329-44f6 and SIP/328-e85a set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.205, port 5060 We're at 142.46.202.202 port 10056 Answering/Requesting with root capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 12 lines Reliably Transmitting: INVITE sip:328@10.1.0.205 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4c5a61f1 From: "Nicole Clientcare" ;tag=as52811732 To: ;tag=E104C494-C731B3ED Contact: Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 259 v=0 o=root 8904 8905 IN IP4 10.1.0.213 s=session c=IN IP4 10.1.0.213 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.1.0.205:5060 Sip read: ACK sip:328@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK388eeff8B9C46A07 From: "329" ;tag=FA854A02-E46D187F To: ;tag=as6273cf46 CSeq: 2 ACK Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 11 headers, 0 lines set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.213, port 5060 We're at 142.46.202.202 port 10006 Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 10 lines Reliably Transmitting: INVITE sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK44f1add5;rport From: ;tag=as6273cf46 To: "329" ;tag=FA854A02-E46D187F Contact: Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 209 v=0 o=root 8904 8905 IN IP4 10.1.0.205 s=session c=IN IP4 10.1.0.205 t=0 0 m=audio 2230 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 10.1.0.213:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4c5a61f1 From: "Nicole Clientcare" ;tag=as52811732 To: ;tag=E104C494-C731B3ED CSeq: 103 INVITE Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1134398869 1134398869 IN IP4 10.1.0.205 s=Polycom IP Phone c=IN IP4 10.1.0.205 t=0 0 m=audio 2230 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 11 headers, 8 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.205:2230 Found description format PCMU Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.205, port 5060 Transmitting: ACK sip:328@10.1.0.205 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK75383b02 From: "Nicole Clientcare" ;tag=as52811732 To: ;tag=E104C494-C731B3ED Contact: Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.1.0.205:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK44f1add5;rport From: ;tag=as6273cf46 To: "329" ;tag=FA854A02-E46D187F CSeq: 102 INVITE Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1134398894 1134398894 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 10.1.0.213 t=0 0 m=audio 2228 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 11 headers, 8 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.213:2228 Found description format PCMU Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.213, port 5060 Transmitting: ACK sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0f200de5;rport From: ;tag=as6273cf46 To: "329" ;tag=FA854A02-E46D187F Contact: Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 10.1.0.213:5060 11 headers, 0 lines Dec 12 09:48:31 NOTICE[12125]: chan_sip.c:6831 handle_response: Outbound Registration: Expiry for sphone.vopr.vonage.net is 20 sec (Scheduling reregistration in 15999 ms) Destroying call '27cc5b96406642f8204cf2a065aa4a99@142.46.202.202' Destroying call 'cabc8274-67f0c32e-f49192dd@10.1.0.222' Sip read: 0 headers, 0 lines Sip read: REFER sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKaebacea842AA011 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 6 REFER Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 12 headers, 0 lines Looking for 328 in itd01-sip Looking for 329 in itd01-sip Transmitting (no NAT): SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bKaebacea842AA011 From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 6 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.213:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.213, port 5060 Reliably Transmitting: NOTIFY sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1e298f66 From: "HD/24443/A CUSTOMER" ;tag=as68cc696f To: ;tag=F2B22CDD-B293F366 Contact: Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Event: refer;id=6 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK (no NAT) to 10.1.0.213:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.213, port 5060 Reliably Transmitting: BYE sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4385964a From: "HD/24443/A CUSTOMER" ;tag=as68cc696f To: ;tag=F2B22CDD-B293F366 Contact: Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 104 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.1.0.213:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.213, port 5060 We're at 142.46.202.202 port 10006 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x400 (ilbc) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 13 lines Reliably Transmitting: INVITE sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK667f8d68;rport From: ;tag=as6273cf46 To: "329" ;tag=FA854A02-E46D187F Contact: Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 293 v=0 o=root 8904 8906 IN IP4 142.46.202.202 s=session c=IN IP4 142.46.202.202 t=0 0 m=audio 10006 RTP/AVP 0 18 3 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 10.1.0.213:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1e298f66 From: "HD/24443/A CUSTOMER" ;tag=as68cc696f To: ;tag=F2B22CDD-B293F366 CSeq: 103 NOTIFY Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: Event: refer;id=6 User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Message is NOTIFY Sip read: BYE sip:9058431234@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK1841362c2FE0C2B From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f CSeq: 7 BYE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 10 headers, 0 lines Sending to 10.1.0.213 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.213;branch=z9hG4bK1841362c2FE0C2B From: ;tag=F2B22CDD-B293F366 To: "HD/24443/A CUSTOMER" ;tag=as68cc696f Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 CSeq: 7 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.213:5060 Sip read: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4385964a From: "HD/24443/A CUSTOMER" ;tag=as68cc696f To: ;tag=F2B22CDD-B293F366 CSeq: 104 BYE Call-ID: 3f7b1b045c4ead1e231c578634af242e@142.46.202.202 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 9 headers, 0 lines Message is BYE Destroying call '3f7b1b045c4ead1e231c578634af242e@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK667f8d68;rport From: ;tag=as6273cf46 To: "329" ;tag=FA854A02-E46D187F CSeq: 103 INVITE Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1134398895 1134398895 IN IP4 10.1.0.213 s=Polycom IP Phone c=IN IP4 10.1.0.213 t=0 0 m=audio 2228 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 11 headers, 8 lines Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.1.0.213:2228 Found description format PCMU Found description format telephone-event Capabilities: us - 0x506 (gsm|ulaw|g729|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.213, port 5060 Transmitting: ACK sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0c723a3c;rport From: ;tag=as6273cf46 To: "329" ;tag=FA854A02-E46D187F Contact: Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 10.1.0.213:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.0.213, port 5060 Reliably Transmitting: BYE sip:329@10.1.0.213 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7479c1d3;rport From: ;tag=as6273cf46 To: "329" ;tag=FA854A02-E46D187F Contact: Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 CSeq: 104 BYE User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 10.1.0.213:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7479c1d3;rport From: ;tag=as6273cf46 To: "329" ;tag=FA854A02-E46D187F CSeq: 104 BYE Call-ID: a22e6e6e-4289c950-f0a17b45@10.1.0.213 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 9 headers, 0 lines Destroying call 'a22e6e6e-4289c950-f0a17b45@10.1.0.213' 11 headers, 0 lines 11 headers, 0 lines Sip read: 0 headers, 0 lines Sip read: BYE sip:329@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.205;branch=z9hG4bK5b84639771DB6966 From: ;tag=E104C494-C731B3ED To: "Nicole Clientcare" ;tag=as52811732 CSeq: 1 BYE Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Max-Forwards: 70 Content-Length: 0 10 headers, 0 lines Sending to 10.1.0.205 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.205;branch=z9hG4bK5b84639771DB6966 From: ;tag=E104C494-C731B3ED To: "Nicole Clientcare" ;tag=as52811732 Call-ID: 628bbb357a7f30a0084036796e3f6b53@142.46.202.202 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.205:5060 monitor executing ( nice -n 19 soxmix "/var/spool/asterisk/monitor/agent-329-1134398790-11259-in.WAV" "/var/spool/asterisk/monitor/agent-329-1134398790-11259-out.WAV" "/var/spool/asterisk/monitor/agent-329-1134398790-11259.WAV" && rm -f "/var/spool/asterisk/monitor/agent-329-1134398790-11259-"* ) & ;original caller is lost here. -- Hungup 'IAX2/oce01pbx@216.7.201.43:4569/10' Destroying call '628bbb357a7f30a0084036796e3f6b53@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:328@10.1.0.205 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0b8806f6 From: "asterisk" ;tag=as053bf11c To: Contact: Call-ID: 58aee31109a3046631e22d1562638ad0@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.205:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0b8806f6 From: "asterisk" ;tag=as053bf11c To: ;tag=8CE75C3B-27A786E8 CSeq: 102 OPTIONS Call-ID: 58aee31109a3046631e22d1562638ad0@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '58aee31109a3046631e22d1562638ad0@142.46.202.202'