ernestine*CLI> <--- SIP read from 192.168.255.125:5060 ---> INVITE sip:8581@192.168.255.2:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK65d44db6EEFCB61F From: "Earl Tom" ;tag=43832E73-AD7E74 To: CSeq: 1 INVITE Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 279 v=0 o=- 1208821088 1208821088 IN IP4 192.168.255.125 s=Polycom IP Phone c=IN IP4 192.168.255.125 t=0 0 m=audio 2224 RTP/AVP 9 0 8 18 101 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> ernestine*CLI> --- (14 headers 12 lines) --- ernestine*CLI> Sending to 192.168.255.125 : 5060 (no NAT) ernestine*CLI> Using INVITE request as basis request - 190f42c0-c05dd5d9-29011faa@192.168.255.125 ernestine*CLI> Found peer '8525' ernestine*CLI> <--- Reliably Transmitting (no NAT) to 192.168.255.125:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK65d44db6EEFCB61F;received=192.168.255.125 From: "Earl Tom" ;tag=43832E73-AD7E74 To: ;tag=as103f2725 Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="45f30a70" Content-Length: 0 <------------> ernestine*CLI> Scheduling destruction of SIP dialog '190f42c0-c05dd5d9-29011faa@192.168.255.125' in 32000 ms (Method: INVITE) ernestine*CLI> <--- SIP read from 192.168.255.125:5060 ---> ACK sip:8581@192.168.255.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK65d44db6EEFCB61F From: "Earl Tom" ;tag=43832E73-AD7E74 To: ;tag=as103f2725 CSeq: 1 ACK Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Max-Forwards: 70 Content-Length: 0 <-------------> ernestine*CLI> --- (11 headers 0 lines) --- ernestine*CLI> <--- SIP read from 192.168.255.125:5060 ---> INVITE sip:8581@192.168.255.2:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK7b9e89ed662AD91E From: "Earl Tom" ;tag=43832E73-AD7E74 To: CSeq: 2 INVITE Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="8525", realm="asterisk", nonce="45f30a70", uri="sip:8581@192.168.255.2:5060;user=phone;transport=udp", response="dc259dbc545ac1a278394affae870235", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 279 v=0 o=- 1208821088 1208821088 IN IP4 192.168.255.125 s=Polycom IP Phone c=IN IP4 192.168.255.125 t=0 0 m=audio 2224 RTP/AVP 9 0 8 18 101 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> ernestine*CLI> --- (15 headers 12 lines) --- ernestine*CLI> Sending to 192.168.255.125 : 5060 (no NAT) ernestine*CLI> Using INVITE request as basis request - 190f42c0-c05dd5d9-29011faa@192.168.255.125 ernestine*CLI> Found peer '8525' ernestine*CLI> Found RTP audio format 9 ernestine*CLI> Found RTP audio format 0 ernestine*CLI> Found RTP audio format 8 ernestine*CLI> Found RTP audio format 18 ernestine*CLI> Found RTP audio format 101 ernestine*CLI> Peer audio RTP is at port 192.168.255.125:2224 ernestine*CLI> Found audio description format G722 for ID 9 ernestine*CLI> Found audio description format PCMU for ID 0 ernestine*CLI> Found audio description format PCMA for ID 8 ernestine*CLI> Found audio description format G729 for ID 18 ernestine*CLI> Found audio description format telephone-event for ID 101 ernestine*CLI> Capabilities: us - 0x1004 (ulaw|g722), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing), combined - 0x1004 (ulaw|g722) ernestine*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) ernestine*CLI> Peer audio RTP is at port 192.168.255.125:2224 ernestine*CLI> Looking for 8581 in corp-carrollton (domain 192.168.255.2) ernestine*CLI> list_route: hop: ernestine*CLI> <--- Transmitting (no NAT) to 192.168.255.125:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK7b9e89ed662AD91E;received=192.168.255.125 From: "Earl Tom" ;tag=43832E73-AD7E74 To: Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> ernestine*CLI> -- Executing [8581@corp-carrollton:1] Goto("SIP/8525-afa8a290", "sip-extens-personal-queue|8581|1") in new stack ernestine*CLI> -- Goto (sip-extens-personal-queue,8581,1) ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:1] Gosub("SIP/8525-afa8a290", "initcdr|8581|1") in new stack ernestine*CLI> -- Executing [8581@initcdr:1] GotoIf("SIP/8525-afa8a290", "0 ?unknown:known") in new stack ernestine*CLI> -- Goto (initcdr,8581,3) ernestine*CLI> -- Executing [8581@initcdr:3] NoOp("SIP/8525-afa8a290", ""Callerid not blank"") in new stack ernestine*CLI> -- Executing [8581@initcdr:4] GotoIf("SIP/8525-afa8a290", "1?newcdr:existingcdr") in new stack ernestine*CLI> -- Goto (initcdr,8581,5) ernestine*CLI> -- Executing [8581@initcdr:5] NoOp("SIP/8525-afa8a290", ""Create New CDR record"") in new stack ernestine*CLI> -- Executing [8581@initcdr:6] Set("SIP/8525-afa8a290", "__cdruserfield=1208821092 from 8525 to 8581") in new stack ernestine*CLI> -- Executing [8581@initcdr:7] Set("SIP/8525-afa8a290", "__cdraccountcode=1208821092") in new stack ernestine*CLI> -- Executing [8581@initcdr:8] Return("SIP/8525-afa8a290", "") in new stack ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:2] GotoIf("SIP/8525-afa8a290", "0 ?self:notself") in new stack ernestine*CLI> -- Goto (sip-extens-personal-queue,8581,7) -- Executing [8581@sip-extens-personal-queue:7] NoOp("SIP/8525-afa8a290", ""Call from external party"") in new stack ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:8] Set("SIP/8525-afa8a290", "__cdruserfield=1208821092 from 8525 to 8581 CALLIN8581") in new stack ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:9] GotoIf("SIP/8525-afa8a290", "1 ?in:out") in new stack ernestine*CLI> -- Goto (sip-extens-personal-queue,8581,10) ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:10] NoOp("SIP/8525-afa8a290", ""Logged In - Dial Phone"") in new stack ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:11] GotoIf("SIP/8525-afa8a290", "1)?noalert:alert") in new stack -- Goto (sip-extens-personal-queue,8581,13) ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:13] GotoIf("SIP/8525-afa8a290", "1 ?directcall:queuecall") in new stack ernestine*CLI> -- Goto (sip-extens-personal-queue,8581,14) ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:14] Set("SIP/8525-afa8a290", "__cdruserfield=1208821092 from 8525 to 8581 CALLIN8581 DIAL8581") in new stack -- Executing [8581@sip-extens-personal-queue:15] Set("SIP/8525-afa8a290", "CDR(userfield)=1208821092 from 8525 to 8581 CALLIN8581 DIAL8581") in new stack ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:16] Set("SIP/8525-afa8a290", "CDR(accountcode)=1208821092") in new stack ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:17] Dial("SIP/8525-afa8a290", "SIP/8581|20|wW") in new stack ernestine*CLI> Audio is at 192.168.255.2 port 16624 ernestine*CLI> Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP ernestine*CLI> Reliably Transmitting (no NAT) to 192.168.255.181:5060: INVITE sip:8581@192.168.255.181;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK1e147028;rport From: "Earl Tom" ;tag=as5a856702 To: Contact: Call-ID: 5a3ffc4a355d31544cbe003b7e30691c@192.168.255.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 Apr 2008 23:38:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "Earl Tom" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 242 v=0 o=root 21466 21466 IN IP4 192.168.255.2 s=session c=IN IP4 192.168.255.2 t=0 0 m=audio 16624 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 8581 ernestine*CLI> <--- Transmitting (no NAT) to 192.168.255.125:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK7b9e89ed662AD91E;received=192.168.255.125 From: "Earl Tom" ;tag=43832E73-AD7E74 To: ;tag=as32f64775 Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Remote-Party-ID: "n+" ;party=called;privacy=off;screen=no <------------> ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK1e147028;rport From: "Earl Tom" ;tag=as5a856702 To: ;tag=6FFDB71-592C4B66 CSeq: 102 INVITE Call-ID: 5a3ffc4a355d31544cbe003b7e30691c@192.168.255.2 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Content-Length: 0 <-------------> ernestine*CLI> --- (9 headers 0 lines) --- ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK1e147028;rport From: "Earl Tom" ;tag=as5a856702 To: ;tag=6FFDB71-592C4B66 CSeq: 102 INVITE Call-ID: 5a3ffc4a355d31544cbe003b7e30691c@192.168.255.2 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> ernestine*CLI> --- (10 headers 0 lines) --- ernestine*CLI> -- SIP/8581-0073ff10 is ringing <--- Transmitting (no NAT) to 192.168.255.125:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK7b9e89ed662AD91E;received=192.168.255.125 From: "Earl Tom" ;tag=43832E73-AD7E74 To: ;tag=as32f64775 Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK1e147028;rport From: "Earl Tom" ;tag=as5a856702 To: ;tag=6FFDB71-592C4B66 CSeq: 102 INVITE Call-ID: 5a3ffc4a355d31544cbe003b7e30691c@192.168.255.2 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Content-Type: application/sdp Content-Length: 205 v=0 o=- 1208821102 1208821102 IN IP4 192.168.255.181 s=Polycom IP Phone c=IN IP4 192.168.255.181 t=0 0 m=audio 2226 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> ernestine*CLI> --- (11 headers 9 lines) --- ernestine*CLI> Found RTP audio format 0 ernestine*CLI> Found RTP audio format 101 ernestine*CLI> Peer audio RTP is at port 192.168.255.181:2226 ernestine*CLI> Found audio description format PCMU for ID 0 ernestine*CLI> Found audio description format telephone-event for ID 101 ernestine*CLI> Capabilities: us - 0x1004 (ulaw|g722), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) ernestine*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) ernestine*CLI> Peer audio RTP is at port 192.168.255.181:2226 ernestine*CLI> list_route: hop: ernestine*CLI> set_destination: Parsing for address/port to send to ernestine*CLI> set_destination: set destination to 192.168.255.181, port 5060 Transmitting (no NAT) to 192.168.255.181:5060: ACK sip:8581@192.168.255.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK5ddc5d37;rport From: "Earl Tom" ;tag=as5a856702 To: ;tag=6FFDB71-592C4B66 Contact: Call-ID: 5a3ffc4a355d31544cbe003b7e30691c@192.168.255.2 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ernestine*CLI> -- SIP/8581-0073ff10 answered SIP/8525-afa8a290 ernestine*CLI> Audio is at 192.168.255.2 port 19048 Adding codec 0x4 (ulaw) to SDP Adding codec 0x1000 (g722) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.255.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK7b9e89ed662AD91E;received=192.168.255.125 From: "Earl Tom" ;tag=43832E73-AD7E74 To: ;tag=as32f64775 Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 266 v=0 o=root 21466 21466 IN IP4 192.168.255.2 s=session c=IN IP4 192.168.255.2 t=0 0 m=audio 19048 RTP/AVP 0 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> ernestine*CLI> <--- SIP read from 192.168.255.125:5060 ---> ACK sip:8581@192.168.255.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK2d504286F6CF5C6F From: "Earl Tom" ;tag=43832E73-AD7E74 To: ;tag=as32f64775 CSeq: 2 ACK Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Proxy-Authorization: Digest username="8525", realm="asterisk", nonce="45f30a70", uri="sip:8581@192.168.255.2:5060;user=phone;transport=udp", response="dc259dbc545ac1a278394affae870235", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> ernestine*CLI> --- (12 headers 0 lines) --- ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> BYE sip:8525@192.168.255.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bKfff5b44b835ED320 From: ;tag=6FFDB71-592C4B66 To: "Earl Tom" ;tag=as5a856702 CSeq: 1 BYE Call-ID: 5a3ffc4a355d31544cbe003b7e30691c@192.168.255.2 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Max-Forwards: 70 Content-Length: 0 <-------------> ernestine*CLI> --- (10 headers 0 lines) --- ernestine*CLI> Sending to 192.168.255.181 : 5060 (no NAT) ernestine*CLI> <--- Transmitting (no NAT) to 192.168.255.181:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bKfff5b44b835ED320;received=192.168.255.181 From: ;tag=6FFDB71-592C4B66 To: "Earl Tom" ;tag=as5a856702 Call-ID: 5a3ffc4a355d31544cbe003b7e30691c@192.168.255.2 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> ernestine*CLI> == Spawn extension (sip-extens-personal-queue, 8581, 17) exited non-zero on 'SIP/8525-afa8a290' ernestine*CLI> set_destination: Parsing for address/port to send to ernestine*CLI> set_destination: set destination to 192.168.255.125, port 5060 ernestine*CLI> Reliably Transmitting (no NAT) to 192.168.255.125:5060: BYE sip:8525@192.168.255.125;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK2873dfae;rport From: ;tag=as32f64775 To: "Earl Tom" ;tag=43832E73-AD7E74 Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- ernestine*CLI> Really destroying SIP dialog '5a3ffc4a355d31544cbe003b7e30691c@192.168.255.2' Method: BYE ernestine*CLI> <--- SIP read from 192.168.255.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK2873dfae;rport From: ;tag=as32f64775 To: "Earl Tom" ;tag=43832E73-AD7E74 CSeq: 102 BYE Call-ID: 190f42c0-c05dd5d9-29011faa@192.168.255.125 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Content-Length: 0 <-------------> ernestine*CLI> --- (9 headers 0 lines) --- ernestine*CLI> SIP Response message for INCOMING dialog BYE arrived ernestine*CLI> Really destroying SIP dialog '190f42c0-c05dd5d9-29011faa@192.168.255.125' Method: ACK