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=z9hG4bK13bc09d5747EC5C6 From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: CSeq: 1 INVITE Call-ID: b74b25af-a1682dd0-33a1a069@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=- 1208821334 1208821334 IN IP4 192.168.255.125 s=Polycom IP Phone c=IN IP4 192.168.255.125 t=0 0 m=audio 2232 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 - b74b25af-a1682dd0-33a1a069@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=z9hG4bK13bc09d5747EC5C6;received=192.168.255.125 From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: ;tag=as133dc9a2 Call-ID: b74b25af-a1682dd0-33a1a069@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="2c119764" Content-Length: 0 <------------> ernestine*CLI> Scheduling destruction of SIP dialog 'b74b25af-a1682dd0-33a1a069@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=z9hG4bK13bc09d5747EC5C6 From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: ;tag=as133dc9a2 CSeq: 1 ACK Call-ID: b74b25af-a1682dd0-33a1a069@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=z9hG4bK3815ff84DAD23A7D From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: CSeq: 2 INVITE Call-ID: b74b25af-a1682dd0-33a1a069@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="2c119764", uri="sip:8581@192.168.255.2:5060;user=phone;transport=udp", response="3adb20b96ddbbe50fcbc7a9294956dca", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 279 v=0 o=- 1208821334 1208821334 IN IP4 192.168.255.125 s=Polycom IP Phone c=IN IP4 192.168.255.125 t=0 0 m=audio 2232 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 - b74b25af-a1682dd0-33a1a069@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 192.168.255.125:2232 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 192.168.255.125:2232 ernestine*CLI> Looking for 8581 in corp-carrollton (domain 192.168.255.2) ernestine*CLI> list_route: hop: <--- Transmitting (no NAT) to 192.168.255.125:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK3815ff84DAD23A7D;received=192.168.255.125 From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: Call-ID: b74b25af-a1682dd0-33a1a069@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 <------------> -- Executing [8581@corp-carrollton:1] Goto("SIP/8525-007f8980", "sip-extens-personal-queue|8581|1") in new stack -- Goto (sip-extens-personal-queue,8581,1) -- Executing [8581@sip-extens-personal-queue:1] Gosub("SIP/8525-007f8980", "initcdr|8581|1") in new stack -- Executing [8581@initcdr:1] GotoIf("SIP/8525-007f8980", "0 ?unknown:known") in new stack -- Goto (initcdr,8581,3) -- Executing [8581@initcdr:3] NoOp("SIP/8525-007f8980", ""Callerid not blank"") in new stack -- Executing [8581@initcdr:4] GotoIf("SIP/8525-007f8980", "1?newcdr:existingcdr") in new stack -- Goto (initcdr,8581,5) -- Executing [8581@initcdr:5] NoOp("SIP/8525-007f8980", ""Create New CDR record"") in new stack -- Executing [8581@initcdr:6] Set("SIP/8525-007f8980", "__cdruserfield=1208821339 from 8525 to 8581") in new stack -- Executing [8581@initcdr:7] Set("SIP/8525-007f8980", "__cdraccountcode=1208821339") in new stack -- Executing [8581@initcdr:8] Return("SIP/8525-007f8980", "") in new stack -- Executing [8581@sip-extens-personal-queue:2] GotoIf("SIP/8525-007f8980", "0 ?self:notself") in new stack -- Goto (sip-extens-personal-queue,8581,7) -- Executing [8581@sip-extens-personal-queue:7] NoOp("SIP/8525-007f8980", ""Call from external party"") in new stack -- Executing [8581@sip-extens-personal-queue:8] Set("SIP/8525-007f8980", "__cdruserfield=1208821339 from 8525 to 8581 CALLIN8581") in new stack -- Executing [8581@sip-extens-personal-queue:9] GotoIf("SIP/8525-007f8980", "1 ?in:out") in new stack -- Goto (sip-extens-personal-queue,8581,10) -- Executing [8581@sip-extens-personal-queue:10] NoOp("SIP/8525-007f8980", ""Logged In - Dial Phone"") in new stack -- Executing [8581@sip-extens-personal-queue:11] GotoIf("SIP/8525-007f8980", "1)?noalert:alert") in new stack -- Goto (sip-extens-personal-queue,8581,13) ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:13] GotoIf("SIP/8525-007f8980", "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/8525-007f8980", "__cdruserfield=1208821339 from 8525 to 8581 CALLIN8581 DIAL8581") in new stack ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:15] Set("SIP/8525-007f8980", "CDR(userfield)=1208821339 from 8525 to 8581 CALLIN8581 DIAL8581") in new stack ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:16] Set("SIP/8525-007f8980", "CDR(accountcode)=1208821339") in new stack ernestine*CLI> -- Executing [8581@sip-extens-personal-queue:17] Dial("SIP/8525-007f8980", "SIP/8581|20|wW") in new stack ernestine*CLI> Audio is at 192.168.255.2 port 16862 ernestine*CLI> Adding codec 0x4 (ulaw) to SDP ernestine*CLI> 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=z9hG4bK27eceebe;rport From: "Earl Tom" ;tag=as2cff2e42 To: Contact: Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 Apr 2008 23:42:19 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 21503 21503 IN IP4 192.168.255.2 s=session c=IN IP4 192.168.255.2 t=0 0 m=audio 16862 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 --- ernestine*CLI> -- 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=z9hG4bK3815ff84DAD23A7D;received=192.168.255.125 From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: ;tag=as66617025 Call-ID: b74b25af-a1682dd0-33a1a069@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=z9hG4bK27eceebe;rport From: "Earl Tom" ;tag=as2cff2e42 To: ;tag=D5B08B67-9F98237C CSeq: 102 INVITE Call-ID: 2414cea835b23ce44a5fe9c70752181a@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=z9hG4bK27eceebe;rport From: "Earl Tom" ;tag=as2cff2e42 To: ;tag=D5B08B67-9F98237C CSeq: 102 INVITE Call-ID: 2414cea835b23ce44a5fe9c70752181a@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/8581-007fccb0 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=z9hG4bK3815ff84DAD23A7D;received=192.168.255.125 From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: ;tag=as66617025 Call-ID: b74b25af-a1682dd0-33a1a069@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=z9hG4bK27eceebe;rport From: "Earl Tom" ;tag=as2cff2e42 To: ;tag=D5B08B67-9F98237C CSeq: 102 INVITE Call-ID: 2414cea835b23ce44a5fe9c70752181a@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=- 1208821345 1208821345 IN IP4 192.168.255.181 s=Polycom IP Phone c=IN IP4 192.168.255.181 t=0 0 m=audio 2238 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 192.168.255.181:2238 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 192.168.255.181:2238 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 ernestine*CLI> 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=z9hG4bK4b9bbc3b;rport From: "Earl Tom" ;tag=as2cff2e42 To: ;tag=D5B08B67-9F98237C Contact: Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- ernestine*CLI> -- SIP/8581-007fccb0 answered SIP/8525-007f8980 ernestine*CLI> Audio is at 192.168.255.2 port 12410 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=z9hG4bK3815ff84DAD23A7D;received=192.168.255.125 From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: ;tag=as66617025 Call-ID: b74b25af-a1682dd0-33a1a069@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 21503 21503 IN IP4 192.168.255.2 s=session c=IN IP4 192.168.255.2 t=0 0 m=audio 12410 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=z9hG4bKf9b7a8111E206422 From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: ;tag=as66617025 CSeq: 2 ACK Call-ID: b74b25af-a1682dd0-33a1a069@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="2c119764", uri="sip:8581@192.168.255.2:5060;user=phone;transport=udp", response="3adb20b96ddbbe50fcbc7a9294956dca", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> ernestine*CLI> --- (12 headers 0 lines) --- ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> INVITE sip:8525@192.168.255.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bKafd21a815A1E4D76 From: ;tag=D5B08B67-9F98237C To: "Earl Tom" ;tag=as2cff2e42 CSeq: 1 INVITE Call-ID: 2414cea835b23ce44a5fe9c70752181a@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 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 205 v=0 o=- 1208821345 1208821346 IN IP4 192.168.255.181 s=Polycom IP Phone c=IN IP4 192.168.255.181 t=0 0 m=audio 2238 RTP/AVP 0 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> ernestine*CLI> --- (14 headers 9 lines) --- ernestine*CLI> Sending to 192.168.255.181 : 5060 (no NAT) ernestine*CLI> Found RTP audio format 0 ernestine*CLI> Found RTP audio format 101 ernestine*CLI> Peer audio RTP is at port 192.168.255.181:2238 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 192.168.255.181:2238 ernestine*CLI> <--- Transmitting (no NAT) to 192.168.255.181:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bKafd21a815A1E4D76;received=192.168.255.181 From: ;tag=D5B08B67-9F98237C To: "Earl Tom" ;tag=as2cff2e42 Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> ernestine*CLI> Audio is at 192.168.255.2 port 16862 ernestine*CLI> Adding codec 0x4 (ulaw) to SDP ernestine*CLI> Adding non-codec 0x1 (telephone-event) to SDP ernestine*CLI> <--- Reliably Transmitting (no NAT) to 192.168.255.181:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bKafd21a815A1E4D76;received=192.168.255.181 From: ;tag=D5B08B67-9F98237C To: "Earl Tom" ;tag=as2cff2e42 Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 21503 21504 IN IP4 192.168.255.2 s=session c=IN IP4 192.168.255.2 t=0 0 m=audio 16862 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> ernestine*CLI> -- Started music on hold, class 'default', on SIP/8525-007f8980 ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> ACK sip:8525@192.168.255.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bKe9c9095bB4900B30 From: ;tag=D5B08B67-9F98237C To: "Earl Tom" ;tag=as2cff2e42 CSeq: 1 ACK Call-ID: 2414cea835b23ce44a5fe9c70752181a@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 Max-Forwards: 70 Content-Length: 0 <-------------> ernestine*CLI> --- (11 headers 0 lines) --- ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> INVITE sip:8544@192.168.255.2:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bKc3777ff53E8F04AA From: "Network Ad" ;tag=966E4ADE-79EEA243 To: CSeq: 1 INVITE Call-ID: 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 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=- 1208821348 1208821348 IN IP4 192.168.255.181 s=Polycom IP Phone c=IN IP4 192.168.255.181 t=0 0 m=audio 2240 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 192.168.255.181 : 5060 (no NAT) ernestine*CLI> Using INVITE request as basis request - 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 ernestine*CLI> Found peer '8581' ernestine*CLI> <--- Reliably Transmitting (no NAT) to 192.168.255.181:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bKc3777ff53E8F04AA;received=192.168.255.181 From: "Network Ad" ;tag=966E4ADE-79EEA243 To: ;tag=as1ea90473 Call-ID: 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 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="748b34d9" Content-Length: 0 <------------> ernestine*CLI> Scheduling destruction of SIP dialog '38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181' in 32000 ms (Method: INVITE) ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> ACK sip:8544@192.168.255.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bKc3777ff53E8F04AA From: "Network Ad" ;tag=966E4ADE-79EEA243 To: ;tag=as1ea90473 CSeq: 1 ACK Call-ID: 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 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 192.168.255.181:5060 ---> INVITE sip:8544@192.168.255.2:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bK3724e79899BCC7DD From: "Network Ad" ;tag=966E4ADE-79EEA243 To: CSeq: 2 INVITE Call-ID: 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 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="8581", realm="asterisk", nonce="748b34d9", uri="sip:8544@192.168.255.2:5060;user=phone;transport=udp", response="2feeed22ee5cd9e7291c73f0770f3fe9", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 279 v=0 o=- 1208821348 1208821348 IN IP4 192.168.255.181 s=Polycom IP Phone c=IN IP4 192.168.255.181 t=0 0 m=audio 2240 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 192.168.255.181 : 5060 (no NAT) ernestine*CLI> Using INVITE request as basis request - 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 ernestine*CLI> Found peer '8581' 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 192.168.255.181:2240 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 192.168.255.181:2240 ernestine*CLI> Looking for 8544 in corp-carrollton (domain 192.168.255.2) ernestine*CLI> list_route: hop: <--- Transmitting (no NAT) to 192.168.255.181:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bK3724e79899BCC7DD;received=192.168.255.181 From: "Network Ad" ;tag=966E4ADE-79EEA243 To: Call-ID: 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [8544@corp-carrollton:1] Goto("SIP/8581-0080fe60", "sip-extens-personal-queue|8544|1") in new stack -- Goto (sip-extens-personal-queue,8544,1) -- Executing [8544@sip-extens-personal-queue:1] Gosub("SIP/8581-0080fe60", "initcdr|8544|1") in new stack -- Executing [8544@initcdr:1] GotoIf("SIP/8581-0080fe60", "0 ?unknown:known") in new stack -- Goto (initcdr,8544,3) -- Executing [8544@initcdr:3] NoOp("SIP/8581-0080fe60", ""Callerid not blank"") in new stack -- Executing [8544@initcdr:4] GotoIf("SIP/8581-0080fe60", "1?newcdr:existingcdr") in new stack -- Goto (initcdr,8544,5) -- Executing [8544@initcdr:5] NoOp("SIP/8581-0080fe60", ""Create New CDR record"") in new stack -- Executing [8544@initcdr:6] Set("SIP/8581-0080fe60", "__cdruserfield=1208821352 from 8581 to 8544") in new stack -- Executing [8544@initcdr:7] Set("SIP/8581-0080fe60", "__cdraccountcode=1208821352") in new stack -- Executing [8544@initcdr:8] Return("SIP/8581-0080fe60", "") in new stack -- Executing [8544@sip-extens-personal-queue:2] GotoIf("SIP/8581-0080fe60", "0 ?self:notself") in new stack -- Goto (sip-extens-personal-queue,8544,7) -- Executing [8544@sip-extens-personal-queue:7] NoOp("SIP/8581-0080fe60", ""Call from external party"") in new stack -- Executing [8544@sip-extens-personal-queue:8] Set("SIP/8581-0080fe60", "__cdruserfield=1208821352 from 8581 to 8544 CALLIN8544") in new stack -- Executing [8544@sip-extens-personal-queue:9] GotoIf("SIP/8581-0080fe60", "1 ?in:out") in new stack -- Goto (sip-extens-personal-queue,8544,10) -- Executing [8544@sip-extens-personal-queue:10] NoOp("SIP/8581-0080fe60", ""Logged In - Dial Phone"") in new stack -- Executing [8544@sip-extens-personal-queue:11] GotoIf("SIP/8581-0080fe60", "1)?noalert:alert") in new stack -- Goto (sip-extens-personal-queue,8544,13) -- Executing [8544@sip-extens-personal-queue:13] GotoIf("SIP/8581-0080fe60", "1 ?directcall:queuecall") in new stack -- Goto (sip-extens-personal-queue,8544,14) -- Executing [8544@sip-extens-personal-queue:14] Set("SIP/8581-0080fe60", "__cdruserfield=1208821352 from 8581 to 8544 CALLIN8544 DIAL8544") in new stack -- Executing [8544@sip-extens-personal-queue:15] Set("SIP/8581-0080fe60", "CDR(userfield)=1208821352 from 8581 to 8544 CALLIN8544 DIAL8544") in new stack -- Executing [8544@sip-extens-personal-queue:16] Set("SIP/8581-0080fe60", "CDR(accountcode)=1208821352") in new stack -- Executing [8544@sip-extens-personal-queue:17] Dial("SIP/8581-0080fe60", "SIP/8544|20|wW") in new stack ernestine*CLI> Audio is at 192.168.255.2 port 10280 ernestine*CLI> Adding codec 0x4 (ulaw) to SDP ernestine*CLI> Adding non-codec 0x1 (telephone-event) to SDP ernestine*CLI> Reliably Transmitting (no NAT) to 192.168.255.144:5060: INVITE sip:8544@192.168.255.144;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: Contact: Call-ID: 36305fcd6fb75030289531b626670769@192.168.255.2 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 21 Apr 2008 23:42:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "Tony Chandler" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 242 v=0 o=root 21503 21503 IN IP4 192.168.255.2 s=session c=IN IP4 192.168.255.2 t=0 0 m=audio 10280 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 --- ernestine*CLI> -- Called 8544 ernestine*CLI> <--- Transmitting (no NAT) to 192.168.255.181:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bK3724e79899BCC7DD;received=192.168.255.181 From: "Network Ad" ;tag=966E4ADE-79EEA243 To: ;tag=as20c1fcfb Call-ID: 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Remote-Party-ID: "e+" ;party=called;privacy=off;screen=no <------------> ernestine*CLI> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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 192.168.255.144:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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/8544-00814190 is ringing <--- Transmitting (no NAT) to 192.168.255.181:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bK3724e79899BCC7DD;received=192.168.255.181 From: "Network Ad" ;tag=966E4ADE-79EEA243 To: ;tag=as20c1fcfb Call-ID: 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 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 ---> REFER sip:8525@192.168.255.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bK5d872a517C4D8446 From: ;tag=D5B08B67-9F98237C To: "Earl Tom" ;tag=as2cff2e42 CSeq: 2 REFER Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 <-------------> ernestine*CLI> --- (12 headers 0 lines) --- ernestine*CLI> Call 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 got a SIP call transfer from caller: (REFER)! ernestine*CLI> SIP transfer to extension 8544@corp-carrollton by 8581@192.168.255.2 ernestine*CLI> <--- Transmitting (no NAT) to 192.168.255.181:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bK5d872a517C4D8446;received=192.168.255.181 From: ;tag=D5B08B67-9F98237C To: "Earl Tom" ;tag=as2cff2e42 Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> ernestine*CLI> -- Stopped music on hold on SIP/8525-007f8980 ernestine*CLI> set_destination: Parsing for address/port to send to ernestine*CLI> set_destination: set destination to 192.168.255.181, port 5060 ernestine*CLI> Reliably Transmitting (no NAT) to 192.168.255.181:5060: NOTIFY sip:8581@192.168.255.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK460472c3;rport From: "Earl Tom" ;tag=as2cff2e42 To: ;tag=D5B08B67-9F98237C Contact: Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- 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> Audio is at 192.168.255.2 port 12410 ernestine*CLI> Adding codec 0x4 (ulaw) to SDP ernestine*CLI> Adding codec 0x1000 (g722) to SDP ernestine*CLI> Adding non-codec 0x1 (telephone-event) to SDP ernestine*CLI> Reliably Transmitting (no NAT) to 192.168.255.125:5060: INVITE sip:8525@192.168.255.125;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK33a03139;rport From: ;tag=as66617025 To: "Earl Tom" ;tag=E64B0DBA-CB626403 Contact: Call-ID: b74b25af-a1682dd0-33a1a069@192.168.255.125 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Remote-Party-ID: "8544" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 266 v=0 o=root 21503 21504 IN IP4 192.168.255.2 s=session c=IN IP4 192.168.255.2 t=0 0 m=audio 12410 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> Scheduling destruction of SIP dialog '38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 192.168.255.181:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bK3724e79899BCC7DD;received=192.168.255.181 From: "Network Ad" ;tag=966E4ADE-79EEA243 To: ;tag=as20c1fcfb Call-ID: 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2414cea835b23ce44a5fe9c70752181a@192.168.255.2' in 32000 ms (Method: REFER) == Spawn extension (sip-extens-personal-queue, 8581, 17) exited non-zero on 'SIP/8581-0080fe60' ernestine*CLI> -- SIP/8525-007f8980 requested special control 21, passing it to SIP/8544-00814190 ernestine*CLI> Reliably Transmitting (no NAT) to 192.168.255.144:5060: UPDATE sip:8544@192.168.255.144;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK1ae3968f;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E Contact: Call-ID: 36305fcd6fb75030289531b626670769@192.168.255.2 CSeq: 103 UPDATE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "8525" ;party=calling;privacy=off;screen=no 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=z9hG4bK460472c3;rport From: "Earl Tom" ;tag=as2cff2e42 To: ;tag=D5B08B67-9F98237C CSeq: 103 NOTIFY Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Content-Length: 0 <-------------> ernestine*CLI> --- (10 headers 0 lines) --- ernestine*CLI> SIP Response message for INCOMING dialog NOTIFY arrived 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=z9hG4bK282b72b47A38000 From: ;tag=D5B08B67-9F98237C To: "Earl Tom" ;tag=as2cff2e42 CSeq: 3 BYE Call-ID: 2414cea835b23ce44a5fe9c70752181a@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> Scheduling destruction of SIP dialog '2414cea835b23ce44a5fe9c70752181a@192.168.255.2' in 32000 ms (Method: BYE) 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=z9hG4bK282b72b47A38000;received=192.168.255.181 From: ;tag=D5B08B67-9F98237C To: "Earl Tom" ;tag=as2cff2e42 Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> ernestine*CLI> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK1ae3968f;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 103 UPDATE Call-ID: 36305fcd6fb75030289531b626670769@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> [Apr 21 18:42:32] WARNING[21522]: chan_sip.c:12925 handle_response: Host '192.168.255.144' does not implement 'UPDATE' 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=z9hG4bK33a03139;rport From: ;tag=as66617025 To: "Earl Tom" ;tag=E64B0DBA-CB626403 CSeq: 102 INVITE Call-ID: b74b25af-a1682dd0-33a1a069@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 Content-Type: application/sdp Content-Length: 205 v=0 o=- 1208821334 1208821335 IN IP4 192.168.255.125 s=Polycom IP Phone c=IN IP4 192.168.255.125 t=0 0 m=audio 2232 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 192.168.255.125:2232 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 192.168.255.125:2232 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> Transmitting (no NAT) to 192.168.255.125:5060: ACK sip:8525@192.168.255.125;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK3cc553bb;rport From: ;tag=as66617025 To: "Earl Tom" ;tag=E64B0DBA-CB626403 Contact: Call-ID: b74b25af-a1682dd0-33a1a069@192.168.255.125 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> ACK sip:8544@192.168.255.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.181;branch=z9hG4bK3724e79899BCC7DD From: "Network Ad" ;tag=966E4ADE-79EEA243 To: ;tag=as20c1fcfb CSeq: 2 ACK Call-ID: 38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181 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="8581", realm="asterisk", nonce="748b34d9", uri="sip:8544@192.168.255.2:5060;user=phone;transport=udp", response="2feeed22ee5cd9e7291c73f0770f3fe9", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> ernestine*CLI> --- (12 headers 0 lines) --- ernestine*CLI> <--- SIP read from 192.168.255.125:5060 ---> BYE sip:8581@192.168.255.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK4c8a332bFBCD216C From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: ;tag=as66617025 CSeq: 3 BYE Call-ID: b74b25af-a1682dd0-33a1a069@192.168.255.125 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Proxy-Authorization: Digest username="8525", realm="asterisk", nonce="2c119764", uri="sip:8581@192.168.255.2:5060;user=phone;transport=udp", response="df980b86b714d2ba1964c35d1a26b865", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> ernestine*CLI> --- (11 headers 0 lines) --- ernestine*CLI> Sending to 192.168.255.125 : 5060 (no NAT) ernestine*CLI> <--- Transmitting (no NAT) to 192.168.255.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.125;branch=z9hG4bK4c8a332bFBCD216C;received=192.168.255.125 From: "Earl Tom" ;tag=E64B0DBA-CB626403 To: ;tag=as66617025 Call-ID: b74b25af-a1682dd0-33a1a069@192.168.255.125 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> ernestine*CLI> Scheduling destruction of SIP dialog '36305fcd6fb75030289531b626670769@192.168.255.2' in 32000 ms (Method: INVITE) ernestine*CLI> Reliably Transmitting (no NAT) to 192.168.255.144:5060: CANCEL sip:8544@192.168.255.144;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK1ae3968f;rport From: "Tony Chandler" ;tag=as4211c4d7 To: Call-ID: 36305fcd6fb75030289531b626670769@192.168.255.2 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- ernestine*CLI> Scheduling destruction of SIP dialog '36305fcd6fb75030289531b626670769@192.168.255.2' in 32000 ms (Method: INVITE) ernestine*CLI> == Spawn extension (sip-extens-personal-queue, 8544, 17) exited non-zero on 'SIP/8525-007f8980' ernestine*CLI> Really destroying SIP dialog 'b74b25af-a1682dd0-33a1a069@192.168.255.125' Method: BYE ernestine*CLI> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK1ae3968f;rport From: "Tony Chandler" ;tag=as4211c4d7 To: CSeq: 102 CANCEL Call-ID: 36305fcd6fb75030289531b626670769@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> -- Got SIP response 500 "Internal Server Error" back from 192.168.255.144 ernestine*CLI> Really destroying SIP dialog '36305fcd6fb75030289531b626670769@192.168.255.2' Method: INVITE ernestine*CLI> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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=- 1208821359 1208821359 IN IP4 192.168.255.144 s=Polycom IP Phone c=IN IP4 192.168.255.144 t=0 0 m=audio 2228 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> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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=- 1208821359 1208821359 IN IP4 192.168.255.144 s=Polycom IP Phone c=IN IP4 192.168.255.144 t=0 0 m=audio 2228 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> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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=- 1208821359 1208821359 IN IP4 192.168.255.144 s=Polycom IP Phone c=IN IP4 192.168.255.144 t=0 0 m=audio 2228 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> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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=- 1208821359 1208821359 IN IP4 192.168.255.144 s=Polycom IP Phone c=IN IP4 192.168.255.144 t=0 0 m=audio 2228 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> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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=- 1208821359 1208821359 IN IP4 192.168.255.144 s=Polycom IP Phone c=IN IP4 192.168.255.144 t=0 0 m=audio 2228 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> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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=- 1208821359 1208821359 IN IP4 192.168.255.144 s=Polycom IP Phone c=IN IP4 192.168.255.144 t=0 0 m=audio 2228 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> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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=- 1208821359 1208821359 IN IP4 192.168.255.144 s=Polycom IP Phone c=IN IP4 192.168.255.144 t=0 0 m=audio 2228 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> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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=- 1208821359 1208821359 IN IP4 192.168.255.144 s=Polycom IP Phone c=IN IP4 192.168.255.144 t=0 0 m=audio 2228 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> <--- SIP read from 192.168.255.144:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK6ec48130;rport From: "Tony Chandler" ;tag=as4211c4d7 To: ;tag=9E38B295-E0A48C1E CSeq: 102 INVITE Call-ID: 36305fcd6fb75030289531b626670769@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=- 1208821359 1208821359 IN IP4 192.168.255.144 s=Polycom IP Phone c=IN IP4 192.168.255.144 t=0 0 m=audio 2228 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> Really destroying SIP dialog '38a6a64f-e7f3e1e4-c8a6a469@192.168.255.181' Method: ACK ernestine*CLI> set_destination: Parsing for address/port to send to ernestine*CLI> set_destination: set destination to 192.168.255.181, port 5060 ernestine*CLI> Reliably Transmitting (no NAT) to 192.168.255.181:5060: BYE sip:8581@192.168.255.181 SIP/2.0 Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK5b82b0d6;rport From: "Earl Tom" ;tag=as2cff2e42 To: ;tag=D5B08B67-9F98237C Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- ernestine*CLI> Scheduling destruction of SIP dialog '2414cea835b23ce44a5fe9c70752181a@192.168.255.2' in 32000 ms (Method: BYE) ernestine*CLI> <--- SIP read from 192.168.255.181:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.255.2:5060;branch=z9hG4bK5b82b0d6;rport From: "Earl Tom" ;tag=as2cff2e42 To: ;tag=D5B08B67-9F98237C CSeq: 104 BYE Call-ID: 2414cea835b23ce44a5fe9c70752181a@192.168.255.2 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.0.1.0032 Content-Length: 0 <-------------> ernestine*CLI> --- (8 headers 0 lines) --- ernestine*CLI> SIP Response message for INCOMING dialog BYE arrived ernestine*CLI> Really destroying SIP dialog '2414cea835b23ce44a5fe9c70752181a@192.168.255.2' Method: BYE