keewi*CLI> sip set debug peer 107 SIP Debugging Enabled for IP: 192.168.10.7:5060 -- Executing [107@WholeExtensions:1] Goto("SIP/106-0823bac0", "dial-local|107|1") in new stack -- Goto (dial-local,107,1) -- Executing [107@dial-local:1] Set("SIP/106-0823bac0", "GLOBAL(DIALEDNUMBER)=107") in new stack == Setting global variable 'DIALEDNUMBER' to '107' -- Executing [107@dial-local:2] Set("SIP/106-0823bac0", "VoiceMail=u") in new stack -- Executing [107@dial-local:3] Playback("SIP/106-0823bac0", "2tai-searchcalledparty") in new stack -- Playing '2tai-searchcalledparty' (language 'en') -- Executing [107@dial-local:4] Macro("SIP/106-0823bac0", "rec|") in new stack -- Executing [s@macro-rec:1] GotoIf("SIP/106-0823bac0", "1?end") in new stack -- Goto (macro-rec,s,5) -- Executing [s@macro-rec:5] MacroExit("SIP/106-0823bac0", "") in new stack -- Executing [107@dial-local:5] GotoIf("SIP/106-0823bac0", "0?extendedVM") in new stack -- Executing [107@dial-local:6] Dial("SIP/106-0823bac0", "SIP/107|40|rTt") in new stack Audio is at 192.168.10.250 port 6992 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.10.7:5060: INVITE sip:DH-Poste1@192.168.10.7 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.250:5060;branch=z9hG4bK056dbdfb;rport From: "Tootai" ;tag=as2ce004f8 To: Contact: Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 102 INVITE User-Agent: TOOTAi PBX Max-Forwards: 70 Date: Wed, 23 Jan 2008 11:20:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 265 v=0 o=root 7010 7010 IN IP4 192.168.10.250 s=session c=IN IP4 192.168.10.250 t=0 0 m=audio 6992 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 107 keewi*CLI> <--- SIP read from 192.168.10.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.250:5060;branch=z9hG4bK056dbdfb;rport From: "Tootai" ;tag=as2ce004f8 To: Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 102 INVITE User-Agent: TMS320V5000 TI50002.0.8.3 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- keewi*CLI> <--- SIP read from 192.168.10.7:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.10.250:5060;branch=z9hG4bK056dbdfb;rport From: "Tootai" ;tag=as2ce004f8 To: ;tag=ad37c0ef12a9379a Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 102 INVITE User-Agent: TMS320V5000 TI50002.0.8.3 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/107-08241510 is ringing keewi*CLI> <--- SIP read from 192.168.10.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.250:5060;branch=z9hG4bK056dbdfb;rport From: "Tootai" ;tag=as2ce004f8 To: ;tag=ad37c0ef12a9379a Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 102 INVITE User-Agent: TMS320V5000 TI50002.0.8.3 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 211 v=0 o=107 8000 8000 IN IP4 192.168.10.7 s=SIP Call c=IN IP4 192.168.10.7 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 G711U/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.10.7:5004 Found audio description format G711U for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.10.7:5004 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.10.7, port 5060 Transmitting (no NAT) to 192.168.10.7:5060: ACK sip:107@192.168.10.7;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.10.250:5060;branch=z9hG4bK374ea8f1;rport From: "Tootai" ;tag=as2ce004f8 To: ;tag=ad37c0ef12a9379a Contact: Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 102 ACK User-Agent: TOOTAi PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/107-08241510 answered SIP/106-0823bac0 keewi*CLI> <--- SIP read from 192.168.10.7:5060 ---> INVITE sip:101@192.168.10.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.7;branch=z9hG4bKa52e5cc8b5c2167b From: ;tag=ad37c0ef12a9379a To: "Tootai" ;tag=as2ce004f8 Contact: Supported: replaces Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 32744 INVITE User-Agent: TMS320V5000 TI50002.0.8.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 378 v=0 o=107 8000 8001 IN IP4 192.168.10.7 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 5004 RTP/AVP 0 2 8 18 4 9 97 101 a=sendonly a=rtpmap:0 G711U/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 G711A/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/16000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (13 headers 18 lines) --- Sending to 192.168.10.7 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 9 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:5004 Found audio description format G711U for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G711A for ID 8 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format G722 for ID 9 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x1d0d (g723|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 0.0.0.0:5004 <--- Transmitting (no NAT) to 192.168.10.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.7;branch=z9hG4bKa52e5cc8b5c2167b;received=192.168.10.7 From: ;tag=ad37c0ef12a9379a To: "Tootai" ;tag=as2ce004f8 Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 32744 INVITE User-Agent: TOOTAi PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 192.168.10.250 port 6992 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.10.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.7;branch=z9hG4bKa52e5cc8b5c2167b;received=192.168.10.7 From: ;tag=ad37c0ef12a9379a To: "Tootai" ;tag=as2ce004f8 Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 32744 INVITE User-Agent: TOOTAi PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 7010 7011 IN IP4 192.168.10.250 s=session c=IN IP4 192.168.10.250 t=0 0 m=audio 6992 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on SIP/106-0823bac0 keewi*CLI> <--- SIP read from 192.168.10.7:5060 ---> ACK sip:101@192.168.10.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.7;branch=z9hG4bK9f8fafc2904d5f56 From: ;tag=ad37c0ef12a9379a To: "Tootai" ;tag=as2ce004f8 Contact: Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 32744 ACK User-Agent: TMS320V5000 TI50002.0.8.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- keewi*CLI> <--- SIP read from 192.168.10.7:5060 ---> INVITE sip:101@192.168.10.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.7;branch=z9hG4bK698bfe2669ea7c85 From: ;tag=ad37c0ef12a9379a To: "Tootai" ;tag=as2ce004f8 Contact: Supported: replaces Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 32745 INVITE User-Agent: TMS320V5000 TI50002.0.8.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 383 v=0 o=107 8000 8002 IN IP4 192.168.10.7 s=SIP Call c=IN IP4 192.168.10.7 t=0 0 m=audio 5004 RTP/AVP 0 2 8 18 4 9 97 101 a=sendrecv a=rtpmap:0 G711U/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 G711A/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:9 G722/16000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (13 headers 18 lines) --- Sending to 192.168.10.7 : 5060 (no NAT) Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 9 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 192.168.10.7:5004 Found audio description format G711U for ID 0 Found audio description format G726-32 for ID 2 Found audio description format G711A for ID 8 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format G722 for ID 9 Found audio description format iLBC for ID 97 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x1d0d (g723|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.10.7:5004 <--- Transmitting (no NAT) to 192.168.10.7:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.10.7;branch=z9hG4bK698bfe2669ea7c85;received=192.168.10.7 From: ;tag=ad37c0ef12a9379a To: "Tootai" ;tag=as2ce004f8 Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 32745 INVITE User-Agent: TOOTAi PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 192.168.10.250 port 6992 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.10.7:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.7;branch=z9hG4bK698bfe2669ea7c85;received=192.168.10.7 From: ;tag=ad37c0ef12a9379a To: "Tootai" ;tag=as2ce004f8 Call-ID: 3324123f78a42cc615c3404779cb6835@192.168.10.250 CSeq: 32745 INVITE User-Agent: TOOTAi PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 265 v=0 o=root 7010 7012 IN IP4 192.168.10.250 s=session c=IN IP4 192.168.10.250 t=0 0 m=audio 6992 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - -] Hangup("SIP/106-0823bac0", "") in new stack == Spawn extension (dial-local, h, 1) exited non-zero on 'SIP/106-0823bac0' keewi*CLI>