webdevel*CLI> sip show peer xyz011101 webdevel*CLI> * Name : xyz011101 Secret : MD5Secret : Context : xyz Subscr.Cont. : xyz Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : 12128121210 LastMsgsSent : 32767/65535 Call limit : 2 Dynamic : Yes Callerid : "111v1" <> MaxCallBR : 384 kbps Expire : 341 Insecure : no Nat : Always ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 66.114.80.25 Port 1026 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: xyz011101 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing: No Status : Unmonitored Useragent : PolycomSoundPointIP-SPIP_501-UA/2.1.3.0028 Reg. Contact : sip:xyz011101@192.168.0.250 webdevel*CLI> sip show peer xyz010001 webdevel*CLI> * Name : xyz010001 Secret : MD5Secret : Context : xyz Subscr.Cont. : xyz Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 100@xyz VM Extension : 12128121210 LastMsgsSent : 2/0 Call limit : 2 Dynamic : Yes Callerid : "" <100> MaxCallBR : 384 kbps Expire : 168 Insecure : no Nat : Always ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 66.114.80.25 Port 1024 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: xyz010001 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing: No Status : Unmonitored Useragent : CSCO/7 Reg. Contact : sip:xyz010001@192.168.0.217:5060 webdevel*CLI> <--- SIP read from 66.114.80.25:1026 ---> INVITE sip:100@pbxtest.acecape.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKbf9469c4D544BD31 From: "poly work" ;tag=E6C754E2-B3493945 To: CSeq: 1 INVITE Call-ID: b8a2150e-d6171618-4b9f465b@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.3.0028 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1220473155 1220473155 IN IP4 192.168.0.250 s=Polycom IP Phone c=IN IP4 192.168.0.250 t=0 0 m=audio 2242 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 11 lines) --- webdevel*CLI> Sending to 66.114.80.25 : 1026 (NAT) webdevel*CLI> Using INVITE request as basis request - b8a2150e-d6171618-4b9f465b@192.168.0.250 webdevel*CLI> <--- Reliably Transmitting (NAT) to 66.114.80.25:1026 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKbf9469c4D544BD31;received=66.114.80.25 From: "poly work" ;tag=E6C754E2-B3493945 To: ;tag=as04c315ee Call-ID: b8a2150e-d6171618-4b9f465b@192.168.0.250 CSeq: 1 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="acepbx.com", nonce="60940ec3" Content-Length: 0 <------------> webdevel*CLI> Scheduling destruction of SIP dialog 'b8a2150e-d6171618-4b9f465b@192.168.0.250' in 32000 ms (Method: INVITE) webdevel*CLI> Found user 'xyz011101' webdevel*CLI> <--- SIP read from 66.114.80.25:1026 ---> ACK sip:100@pbxtest.acecape.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKbf9469c4D544BD31 From: "poly work" ;tag=E6C754E2-B3493945 To: ;tag=as04c315ee CSeq: 1 ACK Call-ID: b8a2150e-d6171618-4b9f465b@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.3.0028 Max-Forwards: 70 Content-Length: 0 <-------------> webdevel*CLI> --- (11 headers 0 lines) --- webdevel*CLI> <--- SIP read from 66.114.80.25:1026 ---> INVITE sip:100@pbxtest.acecape.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKfdd1fdef9225B96C From: "poly work" ;tag=E6C754E2-B3493945 To: CSeq: 2 INVITE Call-ID: b8a2150e-d6171618-4b9f465b@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.3.0028 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="xyz011101", realm="acepbx.com", nonce="60940ec3", uri="sip:100@pbxtest.acecape.com:5060;user=phone", response="ecf78c73933e7747cd97cbfffb2595e9", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1220473155 1220473155 IN IP4 192.168.0.250 s=Polycom IP Phone c=IN IP4 192.168.0.250 t=0 0 m=audio 2242 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> webdevel*CLI> --- (15 headers 11 lines) --- webdevel*CLI> Sending to 66.114.80.25 : 1026 (NAT) webdevel*CLI> Using INVITE request as basis request - b8a2150e-d6171618-4b9f465b@192.168.0.250 webdevel*CLI> Found user 'xyz011101' webdevel*CLI> Found RTP audio format 0 webdevel*CLI> Found RTP audio format 8 webdevel*CLI> Found RTP audio format 18 webdevel*CLI> Found RTP audio format 101 webdevel*CLI> Peer audio RTP is at port 192.168.0.250:2242 webdevel*CLI> Found audio description format PCMU for ID 0 webdevel*CLI> Found audio description format PCMA for ID 8 webdevel*CLI> Found audio description format G729 for ID 18 webdevel*CLI> Found audio description format telephone-event for ID 101 webdevel*CLI> Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) webdevel*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) webdevel*CLI> Peer audio RTP is at port 192.168.0.250:2242 webdevel*CLI> Looking for 100 in xyz (domain pbxtest.acecape.com) webdevel*CLI> list_route: hop: webdevel*CLI> <--- Transmitting (NAT) to 66.114.80.25:1026 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKfdd1fdef9225B96C;received=66.114.80.25 From: "poly work" ;tag=E6C754E2-B3493945 To: Call-ID: b8a2150e-d6171618-4b9f465b@192.168.0.250 CSeq: 2 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> webdevel*CLI> -- Executing [100@xyz:1] Set("SIP/xyz011101-081f2830", "TIMEOUT(absolute)=10800") in new stack webdevel*CLI> -- Channel will hangup at 2008-09-04 23:08:01 UTC. webdevel*CLI> -- Executing [100@xyz:2] AGI("SIP/xyz011101-081f2830", "agi://devel.acecape.com") in new stack webdevel*CLI> -- AGI Script Executing Application: (SetCallerPres) Options: (allowed_passed_screen) webdevel*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Ext 100) webdevel*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(num)=Unknown) webdevel*CLI> -- AGI Script Executing Application: (Dial) Options: (SIP/xyz010001|12|Kk) webdevel*CLI> Audio is at 66.114.80.30 port 41416 webdevel*CLI> Adding codec 0x4 (ulaw) to SDP webdevel*CLI> Adding non-codec 0x1 (telephone-event) to SDP webdevel*CLI> Reliably Transmitting (NAT) to 66.114.80.25:5060: INVITE sip:xyz010001@192.168.0.217:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.30:5060;branch=z9hG4bK27e6626c;rport From: "Ext 100" ;tag=as2010ef53 To: Contact: Call-ID: 7da6936437b53bce3e1cca8473c72af1@66.114.80.30 CSeq: 102 INVITE User-Agent: AcePBX Max-Forwards: 70 Date: Thu, 04 Sep 2008 20:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 213 v=0 o=root 19862 19862 IN IP4 66.114.80.30 s=session c=IN IP4 66.114.80.30 t=0 0 m=audio 41416 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- webdevel*CLI> -- Called xyz010001 webdevel*CLI> Retransmitting #1 (NAT) to 66.114.80.25:5060: INVITE sip:xyz010001@192.168.0.217:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.30:5060;branch=z9hG4bK27e6626c;rport From: "Ext 100" ;tag=as2010ef53 To: Contact: Call-ID: 7da6936437b53bce3e1cca8473c72af1@66.114.80.30 CSeq: 102 INVITE User-Agent: AcePBX Max-Forwards: 70 Date: Thu, 04 Sep 2008 20:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 213 v=0 o=root 19862 19862 IN IP4 66.114.80.30 s=session c=IN IP4 66.114.80.30 t=0 0 m=audio 41416 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- webdevel*CLI> Retransmitting #2 (NAT) to 66.114.80.25:5060: INVITE sip:xyz010001@192.168.0.217:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.30:5060;branch=z9hG4bK27e6626c;rport From: "Ext 100" ;tag=as2010ef53 To: Contact: Call-ID: 7da6936437b53bce3e1cca8473c72af1@66.114.80.30 CSeq: 102 INVITE User-Agent: AcePBX Max-Forwards: 70 Date: Thu, 04 Sep 2008 20:08:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 213 v=0 o=root 19862 19862 IN IP4 66.114.80.30 s=session c=IN IP4 66.114.80.30 t=0 0 m=audio 41416 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- webdevel*CLI> <--- SIP read from 66.114.80.25:1026 ---> CANCEL sip:100@pbxtest.acecape.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKfdd1fdef9225B96C From: "poly work" ;tag=E6C754E2-B3493945 To: CSeq: 2 CANCEL Call-ID: b8a2150e-d6171618-4b9f465b@192.168.0.250 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.3.0028 Proxy-Authorization: Digest username="xyz011101", realm="acepbx.com", nonce="60940ec3", uri="sip:100@pbxtest.acecape.com:5060;user=phone", response="ecf78c73933e7747cd97cbfffb2595e9", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> webdevel*CLI> --- (11 headers 0 lines) --- webdevel*CLI> Sending to 66.114.80.25 : 1026 (NAT) webdevel*CLI> <--- Reliably Transmitting (NAT) to 66.114.80.25:1026 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKfdd1fdef9225B96C;received=66.114.80.25 From: "poly work" ;tag=E6C754E2-B3493945 To: ;tag=as286e5a22 Call-ID: b8a2150e-d6171618-4b9f465b@192.168.0.250 CSeq: 2 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> webdevel*CLI> <--- Transmitting (NAT) to 66.114.80.25:1026 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKfdd1fdef9225B96C;received=66.114.80.25 From: "poly work" ;tag=E6C754E2-B3493945 To: ;tag=as286e5a22 Call-ID: b8a2150e-d6171618-4b9f465b@192.168.0.250 CSeq: 2 CANCEL User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> webdevel*CLI> Reliably Transmitting (NAT) to 66.114.80.25:5060: CANCEL sip:xyz010001@192.168.0.217:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.30:5060;branch=z9hG4bK27e6626c;rport From: "Ext 100" ;tag=as2010ef53 To: Call-ID: 7da6936437b53bce3e1cca8473c72af1@66.114.80.30 CSeq: 102 CANCEL User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '7da6936437b53bce3e1cca8473c72af1@66.114.80.30' in 32000 ms (Method: INVITE) webdevel*CLI> == Spawn extension (xyz, 100, 2) exited non-zero on 'SIP/xyz011101-081f2830' webdevel*CLI> -- Executing [h@xyz:1] NoOp("SIP/xyz011101-081f2830", "v2 hangup extension. channel type=SIP") in new stack webdevel*CLI> -- Executing [h@xyz:2] GotoIf("SIP/xyz011101-081f2830", "1?lines_out:+1") in new stack webdevel*CLI> -- Goto (xyz,h,5) webdevel*CLI> -- Executing [h@xyz:5] GotoIf("SIP/xyz011101-081f2830", "1?lines_done:+1") in new stack webdevel*CLI> -- Goto (xyz,h,8) webdevel*CLI> -- Executing [h@xyz:8] NoOp("SIP/xyz011101-081f2830", "") in new stack webdevel*CLI> <--- SIP read from 66.114.80.25:1026 ---> ACK sip:100@pbxtest.acecape.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKfdd1fdef9225B96C From: "poly work" ;tag=E6C754E2-B3493945 To: ;tag=as286e5a22 CSeq: 2 ACK Call-ID: b8a2150e-d6171618-4b9f465b@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.3.0028 Proxy-Authorization: Digest username="xyz011101", realm="acepbx.com", nonce="60940ec3", uri="sip:100@pbxtest.acecape.com:5060;user=phone", response="84db1093c2f6903ab00da5cb8ee18744", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> webdevel*CLI> --- (12 headers 0 lines) --- webdevel*CLI> Really destroying SIP dialog 'b8a2150e-d6171618-4b9f465b@192.168.0.250' Method: ACK <------------> Really destroying SIP dialog '99815781-9a0e8eab-76219e5e@192.168.0.250' Method: SUBSCRIBE webdevel*CLI> Retransmitting #1 (NAT) to 66.114.80.25:5060: CANCEL sip:xyz010001@192.168.0.217:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.30:5060;branch=z9hG4bK27e6626c;rport From: "Ext 100" ;tag=as2010ef53 To: Call-ID: 7da6936437b53bce3e1cca8473c72af1@66.114.80.30 CSeq: 102 CANCEL User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0 --- webdevel*CLI> Really destroying SIP dialog '41128237-1b39ba52-6751ed95@192.168.0.253' Method: REGISTER webdevel*CLI> Retransmitting #2 (NAT) to 66.114.80.25:5060: CANCEL sip:xyz010001@192.168.0.217:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.30:5060;branch=z9hG4bK27e6626c;rport From: "Ext 100" ;tag=as2010ef53 To: Call-ID: 7da6936437b53bce3e1cca8473c72af1@66.114.80.30 CSeq: 102 CANCEL User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0 --- webdevel*CLI> <------------> webdevel*CLI> Really destroying SIP dialog 'dda50778-4dd771ee-1bf32383@192.168.1.250' Method: SUBSCRIBE webdevel*CLI> Retransmitting #3 (NAT) to 66.114.80.25:5060: CANCEL sip:xyz010001@192.168.0.217:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.30:5060;branch=z9hG4bK27e6626c;rport From: "Ext 100" ;tag=as2010ef53 To: Call-ID: 7da6936437b53bce3e1cca8473c72af1@66.114.80.30 CSeq: 102 CANCEL User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0 --- webdevel*CLI> <--- SIP read from 66.114.80.25:1025 ---> <-------------> <------------> webdevel*CLI> Scheduling destruction of SIP dialog '003094c4-4b890004-25c559ea-1d5d9959@192.168.0.217' in 32000 ms (Method: REGISTER) webdevel*CLI> Retransmitting #4 (NAT) to 66.114.80.25:5060: CANCEL sip:xyz010001@192.168.0.217:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.30:5060;branch=z9hG4bK27e6626c;rport From: "Ext 100" ;tag=as2010ef53 To: Call-ID: 7da6936437b53bce3e1cca8473c72af1@66.114.80.30 CSeq: 102 CANCEL User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0 --- webdevel*CLI> Retransmitting #5 (NAT) to 66.114.80.25:5060: CANCEL sip:xyz010001@192.168.0.217:5060 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.30:5060;branch=z9hG4bK27e6626c;rport From: "Ext 100" ;tag=as2010ef53 To: Call-ID: 7da6936437b53bce3e1cca8473c72af1@66.114.80.30 CSeq: 102 CANCEL User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0