-- 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 Reliably Transmitting: OPTIONS sip:4005@192.168.15.99:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK572215b3 From: "asterisk" ;tag=as198a7071 To: Contact: Call-ID: 6bf9980c034ea11b03727ce85ce00aab@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:45:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 142.179.150.29:5060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK572215b3 From: "asterisk" ;tag=as198a7071 To: ;tag=1801156791 Contact: Call-ID: 6bf9980c034ea11b03727ce85ce00aab@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 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 Reliably Transmitting: OPTIONS sip:322@10.1.0.216 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK47e5f721 From: "asterisk" ;tag=as2dfe9756 To: Contact: Call-ID: 1b88d5ac2738cfc3705e835a0a6e8dcc@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.216:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK47e5f721 From: "asterisk" ;tag=as2dfe9756 To: ;tag=C0BAAF1C-23FA48F9 CSeq: 102 OPTIONS Call-ID: 1b88d5ac2738cfc3705e835a0a6e8dcc@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 '1b88d5ac2738cfc3705e835a0a6e8dcc@142.46.202.202' 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=z9hG4bK526739da From: "asterisk" ;tag=as58c7e53c To: Contact: Call-ID: 3fc1bce1774d1f2d348dc9bd2bbb54ee@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:00 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=z9hG4bK526739da From: "asterisk" ;tag=as58c7e53c To: ;tag=BF619C50-D48DBBA7 CSeq: 102 OPTIONS Call-ID: 3fc1bce1774d1f2d348dc9bd2bbb54ee@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 '3fc1bce1774d1f2d348dc9bd2bbb54ee@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=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 Reliably Transmitting: OPTIONS sip:3060@64.26.161.135:10254 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4d128a01 From: "asterisk" ;tag=as01bd49da To: Contact: Call-ID: 03842f7b4cd784c31145fdb54345ec5d@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:05 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=z9hG4bK4d128a01 From: "asterisk" ;tag=as01bd49da To: ;tag=7E23A29-3C2E178E Call-ID: 03842f7b4cd784c31145fdb54345ec5d@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 '03842f7b4cd784c31145fdb54345ec5d@142.46.202.202' Dec 12 09:46:06 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=z9hG4bK45062240 From: ;tag=as779ca0e4 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14777 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=z9hG4bK45062240 From: ;tag=as779ca0e4 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14777 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:46:06 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' 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=z9hG4bK3361f7d6 From: "asterisk" ;tag=as57734eaf To: Contact: Call-ID: 6ccb8bb80d5f5f4855b5dc0f5aabae0b@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:07 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=z9hG4bK3361f7d6 From: "asterisk" ;tag=as57734eaf To: ;tag=31D8CCEB-51D9D92 CSeq: 102 OPTIONS Call-ID: 6ccb8bb80d5f5f4855b5dc0f5aabae0b@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 '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 Dec 12 09:46:14 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=z9hG4bK76403fe9 From: ;tag=as2bbf58bc To: Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9884 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: 9884 REGISTER From: ;tag=as2bbf58bc To: Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK76403fe9 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 09:46:14 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' == 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 Reliably Transmitting: OPTIONS sip:326@10.1.0.173 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK749e651a From: "asterisk" ;tag=as2ad182a7 To: Contact: Call-ID: 2161a53a57fa84a04cddcb7a6a2a51f5@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:16 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=z9hG4bK749e651a From: "asterisk" ;tag=as2ad182a7 To: ;tag=76A90026-BB2EFC13 CSeq: 102 OPTIONS Call-ID: 2161a53a57fa84a04cddcb7a6a2a51f5@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 '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 Reliably Transmitting: OPTIONS sip:310@10.1.0.207 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK29d15478 From: "asterisk" ;tag=as4609f6be To: Contact: Call-ID: 50176b823aceab5e07f321490c780345@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:16 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=z9hG4bK29d15478 From: "asterisk" ;tag=as4609f6be To: ;tag=37AB21E6-CF5D1B13 CSeq: 102 OPTIONS Call-ID: 50176b823aceab5e07f321490c780345@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 '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 Reliably Transmitting: OPTIONS sip:3330@10.1.4.51 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6e0e174a From: "asterisk" ;tag=as466913ac To: Contact: Call-ID: 603c8d890e0fefe7607f9f693837122a@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:19 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=z9hG4bK6e0e174a From: "asterisk" ;tag=as466913ac To: ;tag=B45081DA-8E11870F CSeq: 102 OPTIONS Call-ID: 603c8d890e0fefe7607f9f693837122a@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 '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 Reliably Transmitting: OPTIONS sip:26912@10.1.0.219 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6d221b18 From: "asterisk" ;tag=as4aec9f2d To: Contact: Call-ID: 0bc916e3640fe348742658c84697c241@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:22 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=z9hG4bK6d221b18 From: "asterisk" ;tag=as4aec9f2d To: ;tag=E35F5477-6B427422 CSeq: 102 OPTIONS Call-ID: 0bc916e3640fe348742658c84697c241@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 '0bc916e3640fe348742658c84697c241@142.46.202.202' Dec 12 09:46:22 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=z9hG4bK7e4a9033 From: ;tag=as2c1764eb To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14778 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=z9hG4bK7e4a9033 From: ;tag=as2c1764eb To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14778 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:46:22 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' Sip read: 0 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:26911@64.26.161.135:10254 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK59320bef From: "asterisk" ;tag=as4a0a7863 To: Contact: Call-ID: 1864cea57534092a180696b611ab57fc@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:29 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=z9hG4bK59320bef From: "asterisk" ;tag=as4a0a7863 To: ;tag=42B88A62-70FBC783 Call-ID: 1864cea57534092a180696b611ab57fc@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 '1864cea57534092a180696b611ab57fc@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:4001@192.168.1.50 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK343887e8 From: "asterisk" ;tag=as301a9cbf To: Contact: Call-ID: 49a9ee0124df51947f4174bd695f6c98@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 142.163.161.158:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK343887e8 From: "asterisk" ;tag=as301a9cbf To: ;tag=B580CAA6-76D7676D CSeq: 102 OPTIONS Call-ID: 49a9ee0124df51947f4174bd695f6c98@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Content-Length: 0 10 headers, 0 lines 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 Reliably Transmitting: OPTIONS sip:64.26.157.251 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK43d04d95 From: "asterisk" ;tag=as07bdc4a6 To: Contact: Call-ID: 6c705b601bed8c4b5763101901615e98@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 64.26.157.251:5060 Sip read: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK43d04d95 From: "asterisk" ;tag=as07bdc4a6 To: ;tag=0-fb5c9a1d Call-ID: 6c705b601bed8c4b5763101901615e98@142.46.202.202 CSeq: 102 OPTIONS Content-Length: 0 7 headers, 0 lines Destroying call '6c705b601bed8c4b5763101901615e98@142.46.202.202' -- Attempting native bridge of Zap/2-1 and Zap/3-1 -- Executing NoOp("Local/8665187@fab01-sip-55a2,2", "20051212-094637 fab01-internal call complete") in new stack -- Executing Hangup("Local/8665187@fab01-sip-55a2,2", "") in new stack Dec 12 09:46:38 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=z9hG4bK76358fee From: ;tag=as758b9dcb To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14779 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=z9hG4bK76358fee From: ;tag=as758b9dcb To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14779 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:46:38 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 09:46:38 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=z9hG4bK62ab8c34 From: ;tag=as1b2b7906 To: Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9885 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: 9885 REGISTER From: ;tag=as1b2b7906 To: Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK62ab8c34 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 09:46:38 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' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:198.65.166.131 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK502b9b05 From: "asterisk" ;tag=as3afd2ed5 To: Contact: Call-ID: 44fbce1b75dc165733eda363316c297b@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 198.65.166.131:5060 Sip read: SIP/2.0 404 We could not complete your call as Dialed. Try again. Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK502b9b05 From: "asterisk" ;tag=as3afd2ed5 To: ;tag=21a483426c2cd5d9b85bffe6bba40a2e.d254 Call-ID: 44fbce1b75dc165733eda363316c297b@142.46.202.202 CSeq: 102 OPTIONS Content-Length: 0 7 headers, 0 lines Destroying call '44fbce1b75dc165733eda363316c297b@142.46.202.202' Sip read: 0 headers, 0 lines 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3069@10.1.0.219 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7fb14cc6 From: "asterisk" ;tag=as73aa01b0 To: Contact: Call-ID: 2ca0dcfc09c58a297026e54f2772dbfd@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.219:5060 itd01pbx*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7fb14cc6 From: "asterisk" ;tag=as73aa01b0 To: ;tag=1A4BE6E1-F4F3E214 CSeq: 102 OPTIONS Call-ID: 2ca0dcfc09c58a297026e54f2772dbfd@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 '2ca0dcfc09c58a297026e54f2772dbfd@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3059@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK52b1bbf5 From: "asterisk" ;tag=as1421c2b7 To: Contact: Call-ID: 3a5877e311d9e507319a749c7f5445fe@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 24.42.250.160:5060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK52b1bbf5 From: "asterisk" ;tag=as1421c2b7 To: ;tag=789046058 Contact: Call-ID: 3a5877e311d9e507319a749c7f5445fe@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Destroying call '3a5877e311d9e507319a749c7f5445fe@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:303@10.1.0.203 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3aebe99e From: "asterisk" ;tag=as638df78c To: Contact: Call-ID: 50e654ec406d98b9265b28007d8731e9@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.203:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3aebe99e From: "asterisk" ;tag=as638df78c To: ;tag=1D8F596F-6A35A58 CSeq: 102 OPTIONS Call-ID: 50e654ec406d98b9265b28007d8731e9@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 '50e654ec406d98b9265b28007d8731e9@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:201@10.1.0.209 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0c93164e From: "asterisk" ;tag=as382ce49f To: Contact: Call-ID: 0dbf26a21b6d98e37ce6934548179e85@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 10.1.0.209:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:307@10.1.0.204 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6756c29f From: "asterisk" ;tag=as782c53fc To: Contact: Call-ID: 6326035132f244c61ca62240340c583d@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 10.1.0.204:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:301@10.1.0.214 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2ff5eb6a From: "asterisk" ;tag=as568d980c To: Contact: Call-ID: 72306f845b0924023a80c80a7fef9626@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 10.1.0.214:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:306@10.1.0.222 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK73a39e4b From: "asterisk" ;tag=as0f73192f To: Contact: Call-ID: 7ed36e6e6359b8b5435eb1ca61f971bf@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 10.1.0.222:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:302@10.1.0.202 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK24ec9296 From: "asterisk" ;tag=as47995423 To: Contact: Call-ID: 77e1390f2ce168db27489fca6a11a893@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 10.1.0.202:5060 itd01pbx*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2ff5eb6a From: "asterisk" ;tag=as568d980c To: ;tag=62574FF-D3B20F42 CSeq: 102 OPTIONS Call-ID: 72306f845b0924023a80c80a7fef9626@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 '72306f845b0924023a80c80a7fef9626@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6756c29f From: "asterisk" ;tag=as782c53fc To: ;tag=E339D748-D37C4469 CSeq: 102 OPTIONS Call-ID: 6326035132f244c61ca62240340c583d@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 '6326035132f244c61ca62240340c583d@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0c93164e From: "asterisk" ;tag=as382ce49f To: ;tag=FC75C1DF-4008E9BE CSeq: 102 OPTIONS Call-ID: 0dbf26a21b6d98e37ce6934548179e85@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 '0dbf26a21b6d98e37ce6934548179e85@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK73a39e4b From: "asterisk" ;tag=as0f73192f To: ;tag=E66CB627-966679AE CSeq: 102 OPTIONS Call-ID: 7ed36e6e6359b8b5435eb1ca61f971bf@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 '7ed36e6e6359b8b5435eb1ca61f971bf@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:325@10.1.0.212 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK35447152 From: "asterisk" ;tag=as32087365 To: Contact: Call-ID: 790aa965491af9c12214040c77de17d3@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 10.1.0.212:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK24ec9296 From: "asterisk" ;tag=as47995423 To: ;tag=C7C44208-87E42BD5 CSeq: 102 OPTIONS Call-ID: 77e1390f2ce168db27489fca6a11a893@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 '77e1390f2ce168db27489fca6a11a893@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:308@10.1.0.206 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4709755d From: "asterisk" ;tag=as6fdd538f To: Contact: Call-ID: 65071e08559d93e1758a65955ce85717@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 10.1.0.206:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK35447152 From: "asterisk" ;tag=as32087365 To: ;tag=94FA99-4FCA5A50 CSeq: 102 OPTIONS Call-ID: 790aa965491af9c12214040c77de17d3@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 '790aa965491af9c12214040c77de17d3@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4709755d From: "asterisk" ;tag=as6fdd538f To: ;tag=B37CA4DD-711484D6 CSeq: 102 OPTIONS Call-ID: 65071e08559d93e1758a65955ce85717@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 '65071e08559d93e1758a65955ce85717@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=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' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:320@10.1.0.215 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK73d47ae3 From: "asterisk" ;tag=as61a6a178 To: Contact: Call-ID: 5b417c8808fa9cb93ce78bb740489a90@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.215:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:327@10.1.0.226 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK53ede977 From: "asterisk" ;tag=as642af151 To: Contact: Call-ID: 65ae94387829e17d1f35e2397c94eae4@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.226:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:315@10.1.0.211 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK5caf3167 From: "asterisk" ;tag=as145502dc To: Contact: Call-ID: 2ce52647477f995b2a5e02860826a2cf@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.211:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK53ede977 From: "asterisk" ;tag=as642af151 To: ;tag=E2080849-66830232 CSeq: 102 OPTIONS Call-ID: 65ae94387829e17d1f35e2397c94eae4@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 '65ae94387829e17d1f35e2397c94eae4@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK73d47ae3 From: "asterisk" ;tag=as61a6a178 To: ;tag=EEA76-E07BCAE1 CSeq: 102 OPTIONS Call-ID: 5b417c8808fa9cb93ce78bb740489a90@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 '5b417c8808fa9cb93ce78bb740489a90@142.46.202.202' Sip read: 0 headers, 0 lines Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK5caf3167 From: "asterisk" ;tag=as145502dc To: ;tag=2293CF2A-140902ED CSeq: 102 OPTIONS Call-ID: 2ce52647477f995b2a5e02860826a2cf@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 '2ce52647477f995b2a5e02860826a2cf@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3110@10.1.4.52 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4eac4e03 From: "asterisk" ;tag=as65a02f06 To: Contact: Call-ID: 36ddda165ef1500c6ff5d5421c8c6e4e@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.4.52:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4eac4e03 From: "asterisk" ;tag=as65a02f06 To: ;tag=A7265C17-ACB61490 CSeq: 102 OPTIONS Call-ID: 36ddda165ef1500c6ff5d5421c8c6e4e@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 '36ddda165ef1500c6ff5d5421c8c6e4e@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:202@192.168.15.100:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2350aeff From: "asterisk" ;tag=as25a74429 To: Contact: Call-ID: 1b1d83aa2ff7eb59237bc9044802a9f2@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 24.224.207.64:15060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2350aeff From: "asterisk" ;tag=as25a74429 To: ;tag=3845302247 Contact: Call-ID: 1b1d83aa2ff7eb59237bc9044802a9f2@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Destroying call '1b1d83aa2ff7eb59237bc9044802a9f2@142.46.202.202' Dec 12 09:46:54 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=z9hG4bK421b78be From: ;tag=as18988a21 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14780 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=z9hG4bK421b78be From: ;tag=as18988a21 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14780 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:46:54 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' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:333@10.1.0.218 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1e236e05 From: "asterisk" ;tag=as02d80072 To: Contact: Call-ID: 7e105b1401f98d2f245ef11f71d4ac79@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.218:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:311@10.1.0.208 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3ca94c6d From: "asterisk" ;tag=as123641c8 To: Contact: Call-ID: 2b71ec5a038bd6576a6663967363a722@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.208:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3ca94c6d From: "asterisk" ;tag=as123641c8 To: ;tag=64C129BA-6EB74211 CSeq: 102 OPTIONS Call-ID: 2b71ec5a038bd6576a6663967363a722@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 '2b71ec5a038bd6576a6663967363a722@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1e236e05 From: "asterisk" ;tag=as02d80072 To: ;tag=18ACF202-CE9C39EF CSeq: 102 OPTIONS Call-ID: 7e105b1401f98d2f245ef11f71d4ac79@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 '7e105b1401f98d2f245ef11f71d4ac79@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 Reliably Transmitting: OPTIONS sip:4005@192.168.15.99:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK680347b7 From: "asterisk" ;tag=as71957e14 To: Contact: Call-ID: 7030a9e8673b543a39441a866e4104fd@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:46:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 142.179.150.29:5060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK680347b7 From: "asterisk" ;tag=as71957e14 To: ;tag=301347323 Contact: Call-ID: 7030a9e8673b543a39441a866e4104fd@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Destroying call '7030a9e8673b543a39441a866e4104fd@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:322@10.1.0.216 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3ccf6ad6 From: "asterisk" ;tag=as4ff2074f To: Contact: Call-ID: 5fa624e017da53e462ea05d20b18113a@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.216:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3ccf6ad6 From: "asterisk" ;tag=as4ff2074f To: ;tag=4FD7966D-7956DC46 CSeq: 102 OPTIONS Call-ID: 5fa624e017da53e462ea05d20b18113a@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 '5fa624e017da53e462ea05d20b18113a@142.46.202.202' 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=z9hG4bK666e86d5 From: "asterisk" ;tag=as68e380f4 To: Contact: Call-ID: 1ac27e5b4ee67d8f48ef3f620af32844@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:00 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=z9hG4bK666e86d5 From: "asterisk" ;tag=as68e380f4 To: ;tag=1B77AD1B-AB50DFCA CSeq: 102 OPTIONS Call-ID: 1ac27e5b4ee67d8f48ef3f620af32844@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 '1ac27e5b4ee67d8f48ef3f620af32844@142.46.202.202' == 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' Dec 12 09:47:02 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=z9hG4bK5cb02bf2 From: ;tag=as1c65522a To: Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9886 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: 9886 REGISTER From: ;tag=as1c65522a To: Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK5cb02bf2 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 09:47:02 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' 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=z9hG4bK6beaf521 From: "asterisk" ;tag=as09322830 To: Contact: Call-ID: 64ca21815454c17b0ec655690420d4b5@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:05 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=z9hG4bK6beaf521 From: "asterisk" ;tag=as09322830 To: ;tag=51135847-203F65F4 Call-ID: 64ca21815454c17b0ec655690420d4b5@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 '64ca21815454c17b0ec655690420d4b5@142.46.202.202' 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=z9hG4bK353f81b5 From: "asterisk" ;tag=as37ebeefe To: Contact: Call-ID: 3feecc49566d6f321f91539b7e937987@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:07 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=z9hG4bK353f81b5 From: "asterisk" ;tag=as37ebeefe To: ;tag=FB65E7E6-6C6C2CD5 CSeq: 102 OPTIONS Call-ID: 3feecc49566d6f321f91539b7e937987@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 '3feecc49566d6f321f91539b7e937987@142.46.202.202' 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=z9hG4bK6e044523 From: "asterisk" ;tag=as7f92ec88 To: Contact: Call-ID: 4fa99ac45ae67c4b52d6c8be3473bc45@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:08 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: 4fa99ac45ae67c4b52d6c8be3473bc45@142.46.202.202 CSeq: 102 OPTIONS From: "asterisk" ;tag=as7f92ec88 To: Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6e044523 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 '4fa99ac45ae67c4b52d6c8be3473bc45@142.46.202.202' Dec 12 09:47:10 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=z9hG4bK4198ddb7 From: ;tag=as1f113253 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14781 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=z9hG4bK4198ddb7 From: ;tag=as1f113253 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14781 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:47:10 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' Sip read: 0 headers, 0 lines 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=z9hG4bK48877468 From: "asterisk" ;tag=as37988200 To: Contact: Call-ID: 6b1716b847757ca866af76ec2c568f9d@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:16 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=z9hG4bK48877468 From: "asterisk" ;tag=as37988200 To: ;tag=9D8BB57F-53B011DC CSeq: 102 OPTIONS Call-ID: 6b1716b847757ca866af76ec2c568f9d@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 '6b1716b847757ca866af76ec2c568f9d@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=z9hG4bK73f2cc47 From: "asterisk" ;tag=as1d01756d To: Contact: Call-ID: 0b0d9f1a0a09c2ad3d559ccb54baf163@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:16 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=z9hG4bK73f2cc47 From: "asterisk" ;tag=as1d01756d To: ;tag=BB7C8035-5CCA960E CSeq: 102 OPTIONS Call-ID: 0b0d9f1a0a09c2ad3d559ccb54baf163@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 '0b0d9f1a0a09c2ad3d559ccb54baf163@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=z9hG4bK36d63f11 From: "asterisk" ;tag=as275e138b To: Contact: Call-ID: 28fdc0f515229113451776161414d7ad@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:19 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=z9hG4bK36d63f11 From: "asterisk" ;tag=as275e138b To: ;tag=FE26800-D70F95CD CSeq: 102 OPTIONS Call-ID: 28fdc0f515229113451776161414d7ad@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 '28fdc0f515229113451776161414d7ad@142.46.202.202' Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK59ab775f75642180 From: "Angus Smith" ;tag=D89B9627-36F56B6E To: CSeq: 371 REGISTER Call-ID: fd368ab3-ecad371d-3624bcf4@192.168.1.104 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="330", realm="asterisk", nonce="12d5c6f6", uri="sip:142.46.202.202:5060", response="519596071ada3036b4b86347a8b54d7b", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.1.104 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK59ab775f75642180;received=72.56.142.31;rport=5060 From: "Angus Smith" ;tag=D89B9627-36F56B6E To: Call-ID: fd368ab3-ecad371d-3624bcf4@192.168.1.104 CSeq: 371 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 72.56.142.31:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK59ab775f75642180;received=72.56.142.31;rport=5060 From: "Angus Smith" ;tag=D89B9627-36F56B6E To: ;tag=as0dc341f4 Call-ID: fd368ab3-ecad371d-3624bcf4@192.168.1.104 CSeq: 371 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="736df4d4" Content-Length: 0 to 72.56.142.31:5060 Scheduling destruction of call 'fd368ab3-ecad371d-3624bcf4@192.168.1.104' in 15000 ms Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bKcc5c16893DBA80BA From: "Angus Smith" ;tag=D89B9627-36F56B6E To: CSeq: 372 REGISTER Call-ID: fd368ab3-ecad371d-3624bcf4@192.168.1.104 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="330", realm="asterisk", nonce="736df4d4", uri="sip:142.46.202.202:5060", response="8d424bb3a6a615c3f21c4e7f0fd5b548", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.1.104 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bKcc5c16893DBA80BA;received=72.56.142.31;rport=5060 From: "Angus Smith" ;tag=D89B9627-36F56B6E To: Call-ID: fd368ab3-ecad371d-3624bcf4@192.168.1.104 CSeq: 372 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 72.56.142.31:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bKcc5c16893DBA80BA;received=72.56.142.31;rport=5060 From: "Angus Smith" ;tag=D89B9627-36F56B6E To: ;tag=as0dc341f4 Call-ID: fd368ab3-ecad371d-3624bcf4@192.168.1.104 CSeq: 372 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: ;expires=3600 Date: Mon, 12 Dec 2005 14:47:20 GMT Content-Length: 0 itd01pbx*CLI> to 72.56.142.31:5060 Scheduling destruction of call 'fd368ab3-ecad371d-3624bcf4@192.168.1.104' in 15000 ms 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=z9hG4bK614e08aa From: "asterisk" ;tag=as7c39e914 To: Contact: Call-ID: 24eece422baf203611cd92ec54a32c16@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:22 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=z9hG4bK614e08aa From: "asterisk" ;tag=as7c39e914 To: ;tag=AB1167D0-5D39308F CSeq: 102 OPTIONS Call-ID: 24eece422baf203611cd92ec54a32c16@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 '24eece422baf203611cd92ec54a32c16@142.46.202.202' 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:330@192.168.1.104 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK75f91598;rport From: "asterisk" ;tag=as5af7d52e To: Contact: Call-ID: 330bae3b4cb273fd5836190873771585@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: 0/0 (NAT) to 72.56.142.31:5060 Scheduling destruction of call '330bae3b4cb273fd5836190873771585@142.46.202.202' in 15000 ms Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK75f91598;rport From: "asterisk" ;tag=as5af7d52e To: ;tag=909C1F94-FE861453 CSeq: 102 NOTIFY Call-ID: 330bae3b4cb273fd5836190873771585@142.46.202.202 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '330bae3b4cb273fd5836190873771585@142.46.202.202' Dec 12 09:47:26 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=z9hG4bK483fd167 From: ;tag=as3907da7b To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14782 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=z9hG4bK483fd167 From: ;tag=as3907da7b To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14782 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:47:26 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 09:47:26 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=z9hG4bK613b4cb5 From: ;tag=as2e7ca3b1 To: Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9887 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: 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: SIP/2.0 200 OK Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9887 REGISTER From: ;tag=as2e7ca3b1 To: Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK613b4cb5 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 09:47:26 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: 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 Reliably Transmitting: OPTIONS sip:26911@64.26.161.135:10254 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7bac552f From: "asterisk" ;tag=as3d441218 To: Contact: Call-ID: 2c1246e15233fe3e0f076a8b588172e7@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:29 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=z9hG4bK7bac552f From: "asterisk" ;tag=as3d441218 To: ;tag=FA324BE8-50D154B1 Call-ID: 2c1246e15233fe3e0f076a8b588172e7@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 '2c1246e15233fe3e0f076a8b588172e7@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:4001@192.168.1.50 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1fad5804 From: "asterisk" ;tag=as4e09a997 To: Contact: Call-ID: 31b077f1053f9c534e2920432ba46db9@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 142.163.161.158:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1fad5804 From: "asterisk" ;tag=as4e09a997 To: ;tag=31851F61-357B8360 CSeq: 102 OPTIONS Call-ID: 31b077f1053f9c534e2920432ba46db9@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Content-Length: 0 10 headers, 0 lines Destroying call '31b077f1053f9c534e2920432ba46db9@142.46.202.202' Sip read: 0 headers, 0 lines Destroying call 'fd368ab3-ecad371d-3624bcf4@192.168.1.104' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:64.26.157.251 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK080d2c04 From: "asterisk" ;tag=as1c2773d0 To: Contact: Call-ID: 1a2012250ca8ad9a71c877b546325906@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 64.26.157.251:5060 Sip read: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK080d2c04 From: "asterisk" ;tag=as1c2773d0 To: ;tag=0-b6b1f6b1 Call-ID: 1a2012250ca8ad9a71c877b546325906@142.46.202.202 CSeq: 102 OPTIONS Content-Length: 0 7 headers, 0 lines Destroying call '1a2012250ca8ad9a71c877b546325906@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:198.65.166.131 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK754fe111 From: "asterisk" ;tag=as4f6b57dc To: Contact: Call-ID: 58643366248685867cb45c1d0a14ab57@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 198.65.166.131:5060 Sip read: SIP/2.0 404 We could not complete your call as Dialed. Try again. Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK754fe111 From: "asterisk" ;tag=as4f6b57dc To: ;tag=21a483426c2cd5d9b85bffe6bba40a2e.25af Call-ID: 58643366248685867cb45c1d0a14ab57@142.46.202.202 CSeq: 102 OPTIONS Content-Length: 0 7 headers, 0 lines Destroying call '58643366248685867cb45c1d0a14ab57@142.46.202.202' Sip read: 0 headers, 0 lines Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50;branch=z9hG4bKbb11d97aA31C53EB From: "4001" ;tag=18698C29-6D75F68 To: CSeq: 823 REGISTER Call-ID: dadc9d35-b7b3b67f-5c1b7b2e@192.168.1.50 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Authorization: Digest username="4001", realm="asterisk", nonce="1aae418c", uri="sip:142.46.202.202:5060", response="45862a4550d216ab3f26c954daa2bca8", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.1.50 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.50;branch=z9hG4bKbb11d97aA31C53EB;received=142.163.161.158;rport=5060 From: "4001" ;tag=18698C29-6D75F68 To: Call-ID: dadc9d35-b7b3b67f-5c1b7b2e@192.168.1.50 CSeq: 823 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 142.163.161.158:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.50;branch=z9hG4bKbb11d97aA31C53EB;received=142.163.161.158;rport=5060 From: "4001" ;tag=18698C29-6D75F68 To: ;tag=as10b8405f Call-ID: dadc9d35-b7b3b67f-5c1b7b2e@192.168.1.50 CSeq: 823 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="445a0824" Content-Length: 0 to 142.163.161.158:5060 Scheduling destruction of call 'dadc9d35-b7b3b67f-5c1b7b2e@192.168.1.50' in 15000 ms Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50;branch=z9hG4bK4af9e6f48A6FB0D5 From: "4001" ;tag=18698C29-6D75F68 To: CSeq: 824 REGISTER Call-ID: dadc9d35-b7b3b67f-5c1b7b2e@192.168.1.50 Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Authorization: Digest username="4001", realm="asterisk", nonce="445a0824", uri="sip:142.46.202.202:5060", response="676a457af39deb7e6b8fb1d6d88f3360", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 192.168.1.50 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.50;branch=z9hG4bK4af9e6f48A6FB0D5;received=142.163.161.158;rport=5060 From: "4001" ;tag=18698C29-6D75F68 To: Call-ID: dadc9d35-b7b3b67f-5c1b7b2e@192.168.1.50 CSeq: 824 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 142.163.161.158:5060 Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.50;branch=z9hG4bK4af9e6f48A6FB0D5;received=142.163.161.158;rport=5060 From: "4001" ;tag=18698C29-6D75F68 To: ;tag=as10b8405f Call-ID: dadc9d35-b7b3b67f-5c1b7b2e@192.168.1.50 CSeq: 824 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: ;expires=3600 Date: Mon, 12 Dec 2005 14:47:42 GMT Content-Length: 0 to 142.163.161.158:5060 Scheduling destruction of call 'dadc9d35-b7b3b67f-5c1b7b2e@192.168.1.50' in 15000 ms Dec 12 09:47:42 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=z9hG4bK55a1f682 From: ;tag=as064b3a18 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14783 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=z9hG4bK55a1f682 From: ;tag=as064b3a18 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14783 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:47:42 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' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3069@10.1.0.219 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK07e80bb5 From: "asterisk" ;tag=as18b419d7 To: Contact: Call-ID: 32c0062e1dd464d85fa0d9cc098439f7@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.219:5060 itd01pbx*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK07e80bb5 From: "asterisk" ;tag=as18b419d7 To: ;tag=6ACE1112-FF5C6139 CSeq: 102 OPTIONS Call-ID: 32c0062e1dd464d85fa0d9cc098439f7@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 '32c0062e1dd464d85fa0d9cc098439f7@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3059@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0745e577 From: "asterisk" ;tag=as51161274 To: Contact: Call-ID: 6faf3c9d36ee1af9739e513542ef8912@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 24.42.250.160:5060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0745e577 From: "asterisk" ;tag=as51161274 To: ;tag=3184420498 Contact: Call-ID: 6faf3c9d36ee1af9739e513542ef8912@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Destroying call '6faf3c9d36ee1af9739e513542ef8912@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:303@10.1.0.203 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2068aad9 From: "asterisk" ;tag=as00ef808a To: Contact: Call-ID: 549af2680ee17500417bcd47075af896@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.203:5060 -- Hungup 'Zap/3-1' -- Hungup 'Zap/2-1' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2068aad9 From: "asterisk" ;tag=as00ef808a To: ;tag=6E793AE4-D52CC6F5 CSeq: 102 OPTIONS Call-ID: 549af2680ee17500417bcd47075af896@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 '549af2680ee17500417bcd47075af896@142.46.202.202' 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:4001@192.168.1.50 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK25440665;rport From: "asterisk" ;tag=as1717c8a2 To: Contact: Call-ID: 7e86bc0a00c72ddf2000af596e35f8a7@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 (NAT) to 142.163.161.158:5060 Scheduling destruction of call '7e86bc0a00c72ddf2000af596e35f8a7@142.46.202.202' in 15000 ms Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK25440665;rport From: "asterisk" ;tag=as1717c8a2 To: ;tag=BC3C201F-C5A125CE CSeq: 102 NOTIFY Call-ID: 7e86bc0a00c72ddf2000af596e35f8a7@142.46.202.202 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Content-Length: 0 10 headers, 0 lines Destroying call '7e86bc0a00c72ddf2000af596e35f8a7@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:301@10.1.0.214 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0ff35afa From: "asterisk" ;tag=as0d888bdc To: Contact: Call-ID: 276fe18d1914056734c3e59826bfd67c@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.214:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:307@10.1.0.204 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK700fa0a5 From: "asterisk" ;tag=as09584ad7 To: Contact: Call-ID: 36855c3701cdd8cd052ec6ab0fc76e8b@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.204:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:201@10.1.0.209 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK39c293f1 From: "asterisk" ;tag=as43671e73 To: Contact: Call-ID: 3ae957b36b55a92b5501251e62593940@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.209:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:306@10.1.0.222 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK667d4d2f From: "asterisk" ;tag=as7154b37f To: Contact: Call-ID: 510f404b29494f930a0ca54d07949c82@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.222:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0ff35afa From: "asterisk" ;tag=as0d888bdc To: ;tag=39153DFE-C358542D CSeq: 102 OPTIONS Call-ID: 276fe18d1914056734c3e59826bfd67c@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 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:302@10.1.0.202 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK53e1fa5c From: "asterisk" ;tag=as472428f1 To: Contact: Call-ID: 72e5887f5b7af1b3738733d62dcee032@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.202:5060 Destroying call '276fe18d1914056734c3e59826bfd67c@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK700fa0a5 From: "asterisk" ;tag=as09584ad7 To: ;tag=361E4C95-4DAA5E9E CSeq: 102 OPTIONS Call-ID: 36855c3701cdd8cd052ec6ab0fc76e8b@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 '36855c3701cdd8cd052ec6ab0fc76e8b@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK39c293f1 From: "asterisk" ;tag=as43671e73 To: ;tag=BF5365DA-553CF9BD CSeq: 102 OPTIONS Call-ID: 3ae957b36b55a92b5501251e62593940@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 '3ae957b36b55a92b5501251e62593940@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK667d4d2f From: "asterisk" ;tag=as7154b37f To: ;tag=70D91B52-313D0061 CSeq: 102 OPTIONS Call-ID: 510f404b29494f930a0ca54d07949c82@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 '510f404b29494f930a0ca54d07949c82@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:325@10.1.0.212 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7cc1493c From: "asterisk" ;tag=as1941292e To: Contact: Call-ID: 5d7d62bc0542db04118140502e8ca307@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.212:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK53e1fa5c From: "asterisk" ;tag=as472428f1 To: ;tag=EE63DA49-665788B2 CSeq: 102 OPTIONS Call-ID: 72e5887f5b7af1b3738733d62dcee032@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 '72e5887f5b7af1b3738733d62dcee032@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7cc1493c From: "asterisk" ;tag=as1941292e To: ;tag=5614540D-F2212894 CSeq: 102 OPTIONS Call-ID: 5d7d62bc0542db04118140502e8ca307@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 '5d7d62bc0542db04118140502e8ca307@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:308@10.1.0.206 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK506efd0a From: "asterisk" ;tag=as4d7d4a96 To: Contact: Call-ID: 12d74ffd1f2cbaec58abb4d605bcd87c@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.206:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK506efd0a From: "asterisk" ;tag=as4d7d4a96 To: ;tag=57CB4E0A-528B0BE7 CSeq: 102 OPTIONS Call-ID: 12d74ffd1f2cbaec58abb4d605bcd87c@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 '12d74ffd1f2cbaec58abb4d605bcd87c@142.46.202.202' Sip read: BYE sip:3236@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.205;branch=z9hG4bK2561ad8fE5E3E93E From: ;tag=176F7F15-D80B40FA To: "HelpDesk/Spare" ;tag=as58514d0e CSeq: 3 BYE Call-ID: 2b09ab871ea7f3a332a26ba20c1d5393@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=z9hG4bK2561ad8fE5E3E93E From: ;tag=176F7F15-D80B40FA To: "HelpDesk/Spare" ;tag=as58514d0e Call-ID: 2b09ab871ea7f3a332a26ba20c1d5393@142.46.202.202 CSeq: 3 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-328-1134398662-11246-in.WAV" "/var/spool/asterisk/monitor/agent-328-1134398662-11246-out.WAV" "/var/spool/asterisk/monitor/agent-328-1134398662-11246.WAV" && rm -f "/var/spool/asterisk/monitor/agent-328-1134398662-11246-"* ) & -- Hungup 'IAX2/oce01pbx@216.7.201.43:4569/1' 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=z9hG4bK7876f2a1 From: "asterisk" ;tag=as43b3d220 To: Contact: Call-ID: 06c06e47678ad7fc6e7b0757643dd103@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.205:5060 Destroying call '2b09ab871ea7f3a332a26ba20c1d5393@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7876f2a1 From: "asterisk" ;tag=as43b3d220 To: ;tag=99214AB3-3B1B3F00 CSeq: 102 OPTIONS Call-ID: 06c06e47678ad7fc6e7b0757643dd103@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 '06c06e47678ad7fc6e7b0757643dd103@142.46.202.202' Dec 12 09:47:50 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=z9hG4bK6f771dca From: ;tag=as48ebb393 To: Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9888 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: 9888 REGISTER From: ;tag=as48ebb393 To: Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK6f771dca 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 09:47:51 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 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.208;branch=z9hG4bK5fec6b56450F40EB From: "311" ;tag=9FC803CD-92370C60 To: CSeq: 445 REGISTER Call-ID: a214b801-f76d472f-b8b357ea@10.1.0.208 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="311", realm="asterisk", nonce="44e67712", uri="sip:142.46.202.202", response="ce03be9cc9c11bf9a1fcc7b0e6dba5cb", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 10.1.0.208 : 5060 (NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.208;branch=z9hG4bK5fec6b56450F40EB From: "311" ;tag=9FC803CD-92370C60 To: Call-ID: a214b801-f76d472f-b8b357ea@10.1.0.208 CSeq: 445 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.208:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.0.208;branch=z9hG4bK5fec6b56450F40EB From: "311" ;tag=9FC803CD-92370C60 To: ;tag=as19119ba4 Call-ID: a214b801-f76d472f-b8b357ea@10.1.0.208 CSeq: 445 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="192580ee" Content-Length: 0 to 10.1.0.208:5060 Scheduling destruction of call 'a214b801-f76d472f-b8b357ea@10.1.0.208' in 15000 ms Sip read: REGISTER sip:142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.208;branch=z9hG4bK4c95832cB35B7369 From: "311" ;tag=9FC803CD-92370C60 To: CSeq: 446 REGISTER Call-ID: a214b801-f76d472f-b8b357ea@10.1.0.208 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="311", realm="asterisk", nonce="192580ee", uri="sip:142.46.202.202", response="ce3c7aa058ddb6f379d58d1dd2c327b1", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 10.1.0.208 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.208;branch=z9hG4bK4c95832cB35B7369 From: "311" ;tag=9FC803CD-92370C60 To: Call-ID: a214b801-f76d472f-b8b357ea@10.1.0.208 CSeq: 446 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.208:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.208;branch=z9hG4bK4c95832cB35B7369 From: "311" ;tag=9FC803CD-92370C60 To: ;tag=as19119ba4 Call-ID: a214b801-f76d472f-b8b357ea@10.1.0.208 CSeq: 446 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: ;expires=3600 Date: Mon, 12 Dec 2005 14:47:51 GMT Content-Length: 0 to 10.1.0.208:5060 Scheduling destruction of call 'a214b801-f76d472f-b8b357ea@10.1.0.208' in 15000 ms 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:327@10.1.0.226 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK04aa5761 From: "asterisk" ;tag=as3df03928 To: Contact: Call-ID: 1309b1a839b420167316df885d0f3268@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.226:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:320@10.1.0.215 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK74dd34f2 From: "asterisk" ;tag=as4b868167 To: Contact: Call-ID: 57c678aa2d642d8516d1852651d2b968@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.215:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:315@10.1.0.211 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2ae25afc From: "asterisk" ;tag=as5ca1a587 To: Contact: Call-ID: 57491d696d3d5979544a2f236a52cf11@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.211:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK04aa5761 From: "asterisk" ;tag=as3df03928 To: ;tag=51418516-DF33CA6B CSeq: 102 OPTIONS Call-ID: 1309b1a839b420167316df885d0f3268@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 '1309b1a839b420167316df885d0f3268@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK74dd34f2 From: "asterisk" ;tag=as4b868167 To: ;tag=22EC00D-6CE8C41C CSeq: 102 OPTIONS Call-ID: 57c678aa2d642d8516d1852651d2b968@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 '57c678aa2d642d8516d1852651d2b968@142.46.202.202' Sip read: 0 headers, 0 lines Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2ae25afc From: "asterisk" ;tag=as5ca1a587 To: ;tag=A22C9F41-8834B054 CSeq: 102 OPTIONS Call-ID: 57491d696d3d5979544a2f236a52cf11@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 '57491d696d3d5979544a2f236a52cf11@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3110@10.1.4.52 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK50e55ee3 From: "asterisk" ;tag=as12ec3b3a To: Contact: Call-ID: 4c3b78d045836b4d43fdc1b32401f17a@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.4.52:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK50e55ee3 From: "asterisk" ;tag=as12ec3b3a To: ;tag=136B14E5-6125836 CSeq: 102 OPTIONS Call-ID: 4c3b78d045836b4d43fdc1b32401f17a@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 '4c3b78d045836b4d43fdc1b32401f17a@142.46.202.202' Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bKd0110e9b1FD9B9DA From: "303" ;tag=808DF6C1-B19DFD3A To: CSeq: 281 REGISTER Call-ID: cea8d355-53b27cfb-8a51ac44@10.1.0.203 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="303", realm="asterisk", nonce="267e3979", uri="sip:142.46.202.202:5060", response="b5cddd766b3fd05ee48d1a96dc57f8eb", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 10.1.0.203 : 5060 (NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bKd0110e9b1FD9B9DA From: "303" ;tag=808DF6C1-B19DFD3A To: Call-ID: cea8d355-53b27cfb-8a51ac44@10.1.0.203 CSeq: 281 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.203:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bKd0110e9b1FD9B9DA From: "303" ;tag=808DF6C1-B19DFD3A To: ;tag=as60b8ffef Call-ID: cea8d355-53b27cfb-8a51ac44@10.1.0.203 CSeq: 281 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: WWW-Authenticate: Digest realm="asterisk", nonce="7371d6ae" Content-Length: 0 to 10.1.0.203:5060 Scheduling destruction of call 'cea8d355-53b27cfb-8a51ac44@10.1.0.203' in 15000 ms Sip read: REGISTER sip:142.46.202.202:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bK5dd9c6614A5BFF0 From: "303" ;tag=808DF6C1-B19DFD3A To: CSeq: 282 REGISTER Call-ID: cea8d355-53b27cfb-8a51ac44@10.1.0.203 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="303", realm="asterisk", nonce="7371d6ae", uri="sip:142.46.202.202:5060", response="778b2a4d9404ca2cdcfacdb6d2da547a", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 12 headers, 0 lines Using latest request as basis request Sending to 10.1.0.203 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bK5dd9c6614A5BFF0 From: "303" ;tag=808DF6C1-B19DFD3A To: Call-ID: cea8d355-53b27cfb-8a51ac44@10.1.0.203 CSeq: 282 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.203:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bK5dd9c6614A5BFF0 From: "303" ;tag=808DF6C1-B19DFD3A To: ;tag=as60b8ffef Call-ID: cea8d355-53b27cfb-8a51ac44@10.1.0.203 CSeq: 282 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: ;expires=3600 Date: Mon, 12 Dec 2005 14:47:53 GMT Content-Length: 0 to 10.1.0.203:5060 Scheduling destruction of call 'cea8d355-53b27cfb-8a51ac44@10.1.0.203' in 15000 ms 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:202@192.168.15.100:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2a3102cb From: "asterisk" ;tag=as57ec8426 To: Contact: Call-ID: 3b4c4a4c6c1b572e38ff983702ae4bc6@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 24.224.207.64:15060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2a3102cb From: "asterisk" ;tag=as57ec8426 To: ;tag=3833995861 Contact: Call-ID: 3b4c4a4c6c1b572e38ff983702ae4bc6@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Destroying call '3b4c4a4c6c1b572e38ff983702ae4bc6@142.46.202.202' 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:303@10.1.0.203 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK44748f1d From: "asterisk" ;tag=as740d43a1 To: Contact: Call-ID: 34b1f9682c7847c241fc09a82a86a44a@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: 0/0 (no NAT) to 10.1.0.203:5060 Scheduling destruction of call '34b1f9682c7847c241fc09a82a86a44a@142.46.202.202' in 15000 ms Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK44748f1d From: "asterisk" ;tag=as740d43a1 To: ;tag=8A9AD326-D0D9A447 CSeq: 102 NOTIFY Call-ID: 34b1f9682c7847c241fc09a82a86a44a@142.46.202.202 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '34b1f9682c7847c241fc09a82a86a44a@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=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 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:311@10.1.0.208 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4bdfde68 From: "asterisk" ;tag=as0765a87a To: Contact: Call-ID: 43047fa0520f818665b0ab7534a3092c@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.208:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:333@10.1.0.218 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7736f183 From: "asterisk" ;tag=as48a4ffcb To: Contact: Call-ID: 2816293f1dc996d6534291280f42ec0e@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.218:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4bdfde68 From: "asterisk" ;tag=as0765a87a To: ;tag=C94D9068-F6A9CDD7 CSeq: 102 OPTIONS Call-ID: 43047fa0520f818665b0ab7534a3092c@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 '43047fa0520f818665b0ab7534a3092c@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7736f183 From: "asterisk" ;tag=as48a4ffcb To: ;tag=74622B24-727C6071 CSeq: 102 OPTIONS Call-ID: 2816293f1dc996d6534291280f42ec0e@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 '2816293f1dc996d6534291280f42ec0e@142.46.202.202' 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 Sip read: SUBSCRIBE sip:303@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bK413e7e4dAD50997C From: "303" ;tag=AEB7405-C638A9E To: ;tag=as6d8c76d6 CSeq: 281 SUBSCRIBE Call-ID: ab471719-cc9acb7f-9b0674e8@10.1.0.203 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Accept: application/simple-message-summary Proxy-Authorization: Digest username="303", realm="asterisk", nonce="4a2bdf5a", uri="sip:303@142.46.202.202", response="b544ebd635c3ca0b5917f0861a65b7fc", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 10.1.0.203 : 5060 (NAT) Found peer '303' Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bK413e7e4dAD50997C From: "303" ;tag=AEB7405-C638A9E To: ;tag=as6d8c76d6 Call-ID: ab471719-cc9acb7f-9b0674e8@10.1.0.203 CSeq: 281 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="0841d8c2" Content-Length: 0 to 10.1.0.203:5060 Scheduling destruction of call 'ab471719-cc9acb7f-9b0674e8@10.1.0.203' in 15000 ms Sip read: SUBSCRIBE sip:303@142.46.202.202 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bK28574a7363CED8F2 From: "303" ;tag=AEB7405-C638A9E To: ;tag=as6d8c76d6 CSeq: 282 SUBSCRIBE Call-ID: ab471719-cc9acb7f-9b0674e8@10.1.0.203 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Accept: application/simple-message-summary Proxy-Authorization: Digest username="303", realm="asterisk", nonce="0841d8c2", uri="sip:303@142.46.202.202", response="9a95a9af24d43e1c9b8b196aa0032681", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 15 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 10.1.0.203 : 5060 (non-NAT) Found peer '303' Looking for 303 in itd01-sip Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.203;branch=z9hG4bK28574a7363CED8F2 From: "303" ;tag=AEB7405-C638A9E To: ;tag=as6d8c76d6 Call-ID: ab471719-cc9acb7f-9b0674e8@10.1.0.203 CSeq: 282 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.1.0.203:5060 Destroying call 'ab471719-cc9acb7f-9b0674e8@10.1.0.203' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:4005@192.168.15.99:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK764cebce From: "asterisk" ;tag=as7162f048 To: Contact: Call-ID: 17cd392b28df9e60244b7d1827d750c3@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:47:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 142.179.150.29:5060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK764cebce From: "asterisk" ;tag=as7162f048 To: ;tag=1286783831 Contact: Call-ID: 17cd392b28df9e60244b7d1827d750c3@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Destroying call '17cd392b28df9e60244b7d1827d750c3@142.46.202.202' 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 Reliably Transmitting: REGISTER sip:sphone.vopr.vonage.net SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK01205ab3 From: ;tag=as672dd194 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14784 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 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:311@10.1.0.208 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3a2b3ba7 From: "asterisk" ;tag=as5bac3343 To: Contact: Call-ID: 2649814d42de55d8406f629d654ad436@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 10.1.0.208:5060 Scheduling destruction of call '2649814d42de55d8406f629d654ad436@142.46.202.202' in 15000 ms Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK01205ab3 From: ;tag=as672dd194 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14784 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:47:58 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' 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 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3a2b3ba7 From: "asterisk" ;tag=as5bac3343 To: ;tag=91879CCE-67482FB5 CSeq: 102 NOTIFY Call-ID: 2649814d42de55d8406f629d654ad436@142.46.202.202 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '2649814d42de55d8406f629d654ad436@142.46.202.202' 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 Reliably Transmitting: OPTIONS sip:322@10.1.0.216 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK21da669f From: "asterisk" ;tag=as1df98274 To: Contact: Call-ID: 7ff28d1f2f357c7012d2284d5306b55f@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 10.1.0.216:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK21da669f From: "asterisk" ;tag=as1df98274 To: ;tag=BF4E61AA-6168DACF CSeq: 102 OPTIONS Call-ID: 7ff28d1f2f357c7012d2284d5306b55f@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 '7ff28d1f2f357c7012d2284d5306b55f@142.46.202.202' 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 Reliably Transmitting: NOTIFY sip:307@10.1.0.204 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK37e03d0f From: "asterisk" ;tag=as05a3e951 To: Contact: Call-ID: 2e3bfae44bff29af3de05f5839a93acb@142.46.202.202 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 44 Messages-Waiting: yes Voice-Message: 3/36 (no NAT) to 10.1.0.204:5060 Scheduling destruction of call '2e3bfae44bff29af3de05f5839a93acb@142.46.202.202' in 15000 ms Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK37e03d0f From: "asterisk" ;tag=as05a3e951 To: ;tag=F21746C7-218C4B60 CSeq: 102 NOTIFY Call-ID: 2e3bfae44bff29af3de05f5839a93acb@142.46.202.202 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '2e3bfae44bff29af3de05f5839a93acb@142.46.202.202' 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: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6a27fe1c From: "asterisk" ;tag=as05871beb To: ;tag=519483D3-FE18A57A CSeq: 102 NOTIFY Call-ID: 6030bd45593a86b24ced4d8c7324ff93@142.46.202.202 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.3.0019 Content-Length: 0 10 headers, 0 lines Destroying call '6030bd45593a86b24ced4d8c7324ff93@142.46.202.202' 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 Reliably Transmitting: OPTIONS sip:26911@64.26.161.135:10254 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK62e97554 From: "asterisk" ;tag=as3b7c858e To: Contact: Call-ID: 6a9cfa5f1332a0a57ceb9df8651c2aff@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:29 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=z9hG4bK62e97554 From: "asterisk" ;tag=as3b7c858e To: ;tag=6727A15E-A74E4C0F Call-ID: 6a9cfa5f1332a0a57ceb9df8651c2aff@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 '6a9cfa5f1332a0a57ceb9df8651c2aff@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:4001@192.168.1.50 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7b0c41f7 From: "asterisk" ;tag=as1cd9658b To: Contact: Call-ID: 7676c61450d99931631320a762ebd37b@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 142.163.161.158:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7b0c41f7 From: "asterisk" ;tag=as1cd9658b To: ;tag=49B3FFA2-6050EBC9 CSeq: 102 OPTIONS Call-ID: 7676c61450d99931631320a762ebd37b@142.46.202.202 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Content-Length: 0 10 headers, 0 lines Destroying call '7676c61450d99931631320a762ebd37b@142.46.202.202' Dec 12 09:48:30 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=z9hG4bK4f899214 From: ;tag=as6629ce57 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14786 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 Destroying call 'aad51016-a64b270c-c2d1844d@10.1.0.204' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK4f899214 From: ;tag=as6629ce57 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14786 REGISTER Contact: ;expires=20 Content-Length: 0 8 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 Reliably Transmitting: OPTIONS sip:64.26.157.251 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2f28806b From: "asterisk" ;tag=as18b32f18 To: Contact: Call-ID: 216f1ed108f298996cbcc9e879ccd4ab@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 64.26.157.251:5060 Sip read: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK2f28806b From: "asterisk" ;tag=as18b32f18 To: ;tag=0-12dc60d5 Call-ID: 216f1ed108f298996cbcc9e879ccd4ab@142.46.202.202 CSeq: 102 OPTIONS Content-Length: 0 7 headers, 0 lines Destroying call '216f1ed108f298996cbcc9e879ccd4ab@142.46.202.202' Dec 12 09:48:39 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=z9hG4bK4d79e23a From: ;tag=as742ca7ab To: Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9890 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: 9890 REGISTER From: ;tag=as742ca7ab To: Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK4d79e23a 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 09:48:39 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' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:198.65.166.131 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK17763602 From: "asterisk" ;tag=as4c506d18 To: Contact: Call-ID: 2b6b4d9809ac6892518bc3712062e780@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 198.65.166.131:5060 Sip read: SIP/2.0 404 We could not complete your call as Dialed. Try again. Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK17763602 From: "asterisk" ;tag=as4c506d18 To: ;tag=21a483426c2cd5d9b85bffe6bba40a2e.d809 Call-ID: 2b6b4d9809ac6892518bc3712062e780@142.46.202.202 CSeq: 102 OPTIONS Content-Length: 0 7 headers, 0 lines Destroying call '2b6b4d9809ac6892518bc3712062e780@142.46.202.202' 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-"* ) & -- Hungup 'IAX2/oce01pbx@216.7.201.43:4569/10' Destroying call '628bbb357a7f30a0084036796e3f6b53@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3069@10.1.0.219 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK69c82134 From: "asterisk" ;tag=as3332a621 To: Contact: Call-ID: 3fad006b032300072d3d291c38f0469a@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.219:5060 itd01pbx*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK69c82134 From: "asterisk" ;tag=as3332a621 To: ;tag=958D25BF-DC60200A CSeq: 102 OPTIONS Call-ID: 3fad006b032300072d3d291c38f0469a@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 '3fad006b032300072d3d291c38f0469a@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3059@192.168.0.100:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK75ff6344 From: "asterisk" ;tag=as73646de4 To: Contact: Call-ID: 1400dc967564c8a232d0753d3f6c2a2e@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 24.42.250.160:5060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK75ff6344 From: "asterisk" ;tag=as73646de4 To: ;tag=2742877853 Contact: Call-ID: 1400dc967564c8a232d0753d3f6c2a2e@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Destroying call '1400dc967564c8a232d0753d3f6c2a2e@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:303@10.1.0.203 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK580299a2 From: "asterisk" ;tag=as60549c62 To: Contact: Call-ID: 38a248673376b28e5491a94c784f48d3@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.203:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK580299a2 From: "asterisk" ;tag=as60549c62 To: ;tag=820A311F-C3437788 CSeq: 102 OPTIONS Call-ID: 38a248673376b28e5491a94c784f48d3@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 '38a248673376b28e5491a94c784f48d3@142.46.202.202' Dec 12 09:48:47 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=z9hG4bK08b4a849 From: ;tag=as75965749 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14787 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=z9hG4bK08b4a849 From: ;tag=as75965749 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14787 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:48:47 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' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:301@10.1.0.214 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0c8fe42c From: "asterisk" ;tag=as7e586ae1 To: Contact: Call-ID: 1d259a01358518cd420499035efbd9b4@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 10.1.0.214:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:307@10.1.0.204 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK662099f4 From: "asterisk" ;tag=as5a27cf38 To: Contact: Call-ID: 1ff0dd0d30884aee1576d22f1fccf321@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 10.1.0.204:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:201@10.1.0.209 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK24352d26 From: "asterisk" ;tag=as6b2c5fb5 To: Contact: Call-ID: 3acbf79b1aadaab615af48cc57f1919c@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 10.1.0.209:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:306@10.1.0.222 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK228ec81d From: "asterisk" ;tag=as1f1a4c7c To: Contact: Call-ID: 1944b99727fb943a77b7f5c2393596a4@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 10.1.0.222:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK662099f4 From: "asterisk" ;tag=as5a27cf38 To: ;tag=7021682C-C1F2DE6D CSeq: 102 OPTIONS Call-ID: 1ff0dd0d30884aee1576d22f1fccf321@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 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:302@10.1.0.202 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0b81ac08 From: "asterisk" ;tag=as245b885f To: Contact: Call-ID: 28fb6edf790cb2ac7256d6ce63c7667a@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 10.1.0.202:5060 Destroying call '1ff0dd0d30884aee1576d22f1fccf321@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0c8fe42c From: "asterisk" ;tag=as7e586ae1 To: ;tag=20E32759-33760A4 CSeq: 102 OPTIONS Call-ID: 1d259a01358518cd420499035efbd9b4@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 '1d259a01358518cd420499035efbd9b4@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK228ec81d From: "asterisk" ;tag=as1f1a4c7c To: ;tag=7687FE31-76CA67A8 CSeq: 102 OPTIONS Call-ID: 1944b99727fb943a77b7f5c2393596a4@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 '1944b99727fb943a77b7f5c2393596a4@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK24352d26 From: "asterisk" ;tag=as6b2c5fb5 To: ;tag=F5868369-41E46540 CSeq: 102 OPTIONS Call-ID: 3acbf79b1aadaab615af48cc57f1919c@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 '3acbf79b1aadaab615af48cc57f1919c@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:325@10.1.0.212 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6d445eb8 From: "asterisk" ;tag=as72a66366 To: Contact: Call-ID: 09b4a7bf0b02cb981359947e22f96156@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 10.1.0.212:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK0b81ac08 From: "asterisk" ;tag=as245b885f To: ;tag=4AB66096-6C49BEEB CSeq: 102 OPTIONS Call-ID: 28fb6edf790cb2ac7256d6ce63c7667a@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 '28fb6edf790cb2ac7256d6ce63c7667a@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6d445eb8 From: "asterisk" ;tag=as72a66366 To: ;tag=AD5E4998-31C94E8B CSeq: 102 OPTIONS Call-ID: 09b4a7bf0b02cb981359947e22f96156@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 '09b4a7bf0b02cb981359947e22f96156@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:308@10.1.0.206 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7a61f6d2 From: "asterisk" ;tag=as353181c0 To: Contact: Call-ID: 3cc023ad6914f23524d0d29f65bb928c@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 10.1.0.206:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7a61f6d2 From: "asterisk" ;tag=as353181c0 To: ;tag=819CD1EB-48E95D3C CSeq: 102 OPTIONS Call-ID: 3cc023ad6914f23524d0d29f65bb928c@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 '3cc023ad6914f23524d0d29f65bb928c@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' == Manager 'localscripts' logged on from 142.46.202.202 == Manager 'localscripts' logged off from 142.46.202.202 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:327@10.1.0.226 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7439eed3 From: "asterisk" ;tag=as53266eca To: Contact: Call-ID: 704a700270e1385255ca0552595f6237@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.226:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:320@10.1.0.215 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6a1935ca From: "asterisk" ;tag=as315d095b To: Contact: Call-ID: 0226d6853d2f781a239e2aea0bc9daeb@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.215:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:315@10.1.0.211 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3c565d3e From: "asterisk" ;tag=as6813dab9 To: Contact: Call-ID: 03a7ebe64627c48d0632324473f25be8@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.211:5060 Sip read: 0 headers, 0 lines Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6a1935ca From: "asterisk" ;tag=as315d095b To: ;tag=4E2941D8-6BB430BB CSeq: 102 OPTIONS Call-ID: 0226d6853d2f781a239e2aea0bc9daeb@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 '0226d6853d2f781a239e2aea0bc9daeb@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK7439eed3 From: "asterisk" ;tag=as53266eca To: ;tag=24461FF-30CAB1D0 CSeq: 102 OPTIONS Call-ID: 704a700270e1385255ca0552595f6237@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 '704a700270e1385255ca0552595f6237@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK3c565d3e From: "asterisk" ;tag=as6813dab9 To: ;tag=9FAD168-74EF12CB CSeq: 102 OPTIONS Call-ID: 03a7ebe64627c48d0632324473f25be8@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 '03a7ebe64627c48d0632324473f25be8@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:3110@10.1.4.52 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1504bc47 From: "asterisk" ;tag=as08d2c538 To: Contact: Call-ID: 26cee25d4555fb18108b816e28f5b8e2@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.4.52:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK1504bc47 From: "asterisk" ;tag=as08d2c538 To: ;tag=D010C203-FD0CC86C CSeq: 102 OPTIONS Call-ID: 26cee25d4555fb18108b816e28f5b8e2@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 '26cee25d4555fb18108b816e28f5b8e2@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:202@192.168.15.100:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK048e456d From: "asterisk" ;tag=as552f3eb6 To: Contact: Call-ID: 327489f7787d62f217a7825b361c75dd@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 24.224.207.64:15060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK048e456d From: "asterisk" ;tag=as552f3eb6 To: ;tag=1177539316 Contact: Call-ID: 327489f7787d62f217a7825b361c75dd@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Destroying call '327489f7787d62f217a7825b361c75dd@142.46.202.202' -- Accepting AUTHENTICATED call from 216.7.201.43, requested format = 4, actual format = 4 -- Executing NoOp("IAX2/oce01pbx@216.7.201.43:4569/7", "20051212-094854 in-oce call for itd from "A CUSTOMER" <9058431234> for s") in new stack -- Executing Goto("IAX2/oce01pbx@216.7.201.43:4569/7", "itd01-main|s|1") in new stack -- Goto (itd01-main,s,1) -- Executing Answer("IAX2/oce01pbx@216.7.201.43:4569/7", "") in new stack -- Executing Wait("IAX2/oce01pbx@216.7.201.43:4569/7", "1") in new stack -- Executing ResponseTimeout("IAX2/oce01pbx@216.7.201.43:4569/7", "45") in new stack -- Set Response Timeout to 45 -- Executing SetMusicOnHold("IAX2/oce01pbx@216.7.201.43:4569/7", "default") in new stack -- Executing BackGround("IAX2/oce01pbx@216.7.201.43:4569/7", "itd/welcome") in new stack -- Playing 'itd/welcome' (language 'en') 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:311@10.1.0.208 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK65452c84 From: "asterisk" ;tag=as42692c2f To: Contact: Call-ID: 3a6323b2019a21f454b0fe637fb91eca@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.208:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:333@10.1.0.218 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6986968e From: "asterisk" ;tag=as73d5aa5f To: Contact: Call-ID: 44d8a33744395b4078793fb83d560629@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.218:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK6986968e From: "asterisk" ;tag=as73d5aa5f To: ;tag=ACF68E06-D2DDA4B3 CSeq: 102 OPTIONS Call-ID: 44d8a33744395b4078793fb83d560629@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 '44d8a33744395b4078793fb83d560629@142.46.202.202' Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK65452c84 From: "asterisk" ;tag=as42692c2f To: ;tag=843E5F20-7D6840EF CSeq: 102 OPTIONS Call-ID: 3a6323b2019a21f454b0fe637fb91eca@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 '3a6323b2019a21f454b0fe637fb91eca@142.46.202.202' 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:4005@192.168.15.99:5060 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK290e0fd1 From: "asterisk" ;tag=as71a95208 To: Contact: Call-ID: 50386c607010669632ca56230a9b9012@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:48:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 142.179.150.29:5060 Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK290e0fd1 From: "asterisk" ;tag=as71a95208 To: ;tag=497774122 Contact: Call-ID: 50386c607010669632ca56230a9b9012@142.46.202.202 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Destroying call '50386c607010669632ca56230a9b9012@142.46.202.202' == CDR updated on IAX2/oce01pbx@216.7.201.43:4569/7 -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/7", "silence/1") in new stack -- Playing 'silence/1' (language 'en') -- Executing SetVar("IAX2/oce01pbx@216.7.201.43:4569/7", "QUEUESOURCE=itd01-main") in new stack -- Executing SetVar("IAX2/oce01pbx@216.7.201.43:4569/7", "CIDPREFIX=HelpDesk") in new stack -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/7", "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 Reliably Transmitting: OPTIONS sip:322@10.1.0.216 SIP/2.0 Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK5b7e6431 From: "asterisk" ;tag=as0c9340da To: Contact: Call-ID: 3f1de0a71a2314bd109ee33303f683df@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:49:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.1.0.216:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.46.202.202:5060;branch=z9hG4bK5b7e6431 From: "asterisk" ;tag=as0c9340da To: ;tag=EB41563-E1A72AE4 CSeq: 102 OPTIONS Call-ID: 3f1de0a71a2314bd109ee33303f683df@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 '3f1de0a71a2314bd109ee33303f683df@142.46.202.202' 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=z9hG4bK64be3607 From: "asterisk" ;tag=as3d47b7b1 To: Contact: Call-ID: 6057771d630387f12c709863308fe37d@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:49:00 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=z9hG4bK64be3607 From: "asterisk" ;tag=as3d47b7b1 To: ;tag=72F9AED9-C8452178 CSeq: 102 OPTIONS Call-ID: 6057771d630387f12c709863308fe37d@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 '6057771d630387f12c709863308fe37d@142.46.202.202' Sip read: 0 headers, 0 lines -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/7", "silence/1") in new stack -- Playing 'silence/1' (language 'en') 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=z9hG4bK0a292033 From: "asterisk" ;tag=as4580226d To: Contact: Call-ID: 2360a5b9527c4e805d484c15627e8661@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:49:02 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=z9hG4bK0a292033 From: "asterisk" ;tag=as4580226d To: ;tag=143023A0-E8A974D5 CSeq: 102 OPTIONS Call-ID: 2360a5b9527c4e805d484c15627e8661@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 '2360a5b9527c4e805d484c15627e8661@142.46.202.202' Dec 12 09:49:03 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=z9hG4bK73c672e7 From: ;tag=as27c19448 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14788 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=z9hG4bK73c672e7 From: ;tag=as27c19448 To: Call-ID: 27cc5b96406642f8204cf2a065aa4a99@142.46.202.202 CSeq: 14788 REGISTER Contact: ;expires=20 Content-Length: 0 8 headers, 0 lines Dec 12 09:49:03 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' -- Executing Goto("IAX2/oce01pbx@216.7.201.43:4569/7", "itd01-queue-helpdesk|s|1") in new stack -- Goto (itd01-queue-helpdesk,s,1) -- Executing Playback("IAX2/oce01pbx@216.7.201.43:4569/7", "silence/1") in new stack -- Playing 'silence/1' (language 'en') Dec 12 09:49:03 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=z9hG4bK0cad541e From: ;tag=as60b95eef To: Call-ID: 3e1558785d0c145a4aac97eb594f1752@142.46.202.202 CSeq: 9891 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: 9891 REGISTER From: ;tag=as60b95eef To: Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK0cad541e 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 09:49:03 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' -- Executing ResponseTimeout("IAX2/oce01pbx@216.7.201.43:4569/7", "5") in new stack -- Set Response Timeout to 5 -- Executing BackGround("IAX2/oce01pbx@216.7.201.43:4569/7", "itd/itd-helpdesk-get_ticket") in new stack -- Playing 'itd/itd-helpdesk-get_ticket' (language 'en') 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=z9hG4bK4c470849 From: "asterisk" ;tag=as0ed0433c To: Contact: Call-ID: 22a5de6f6a0d04dc62541d7c75222cef@142.46.202.202 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Mon, 12 Dec 2005 14:49:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 64.26.161.135:10254