--- obelix*CLI> <--- SIP read from 192.168.162.2:5060 ---> INVITE sip:09981@192.168.119.251:5060;user=phone SIP/2.0 Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER Via:SIP/2.0/UDP 192.168.162.2:5060;rport;branch=z9hG4bK4f80994bf9b1bcd5 From: "9980" ;tag=549b7ff5-704166 To: Call-ID:EA1A-B313-467041664491345C6D4A-801@SipHost CSeq:801 INVITE Contact: Expires:90 Max-Forwards:70 Supported:replaces User-Agent:137 12-37-3612753 Content-Type:application/sdp Content-Length:263 v=0 o=9980 1811361680 1811361680 IN IP4 192.168.162.2 s=Session SDP c=IN IP4 192.168.162.2 t=0 0 m=audio 9000 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 <-------------> --- (14 headers 11 lines) --- Sending to 192.168.162.2 : 5060 (NAT) Using INVITE request as basis request - EA1A-B313-467041664491345C6D4A-801@SipHost Found user '9980' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 [Sep 10 15:00:17] DEBUG[29518]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.162.2:9000 Found description format G729 for ID 18 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x108 (alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.162.2:9000 Looking for 09981 in zebra0 (domain 192.168.119.251) list_route: hop: <--- Transmitting (no NAT) to 192.168.162.2:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.162.2:5060;branch=z9hG4bK4f80994bf9b1bcd5;received=192.168.162.2;rport=5060 From: "9980" ;tag=549b7ff5-704166 To: Call-ID: EA1A-B313-467041664491345C6D4A-801@SipHost CSeq: 801 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [09981@zebra0:1] Set("SIP/9980-08208aa0", "DTMFDIAL=09981") in new stack -- Executing [09981@zebra0:2] Macro("SIP/9980-08208aa0", "clr-set|9980") in new stack -- Executing [s@macro-clr-set:1] GotoIf("SIP/9980-08208aa0", "0?:10") in new stack -- Goto (macro-clr-set,s,10) [Sep 10 15:00:17] DEBUG[29884]: app_macro.c:337 _macro_exec: Executed application: GotoIf -- Executing [s@macro-clr-set:10] SetCallerPres("SIP/9980-08208aa0", "allowed") in new stack [Sep 10 15:00:17] DEBUG[29884]: app_macro.c:337 _macro_exec: Executed application: SetCallerPres [Sep 10 15:00:17] DEBUG[29884]: db.c:197 ast_db_get: Unable to find key '9980' in family 'Prohib' [Sep 10 15:00:17] DEBUG[29884]: func_db.c:70 function_db_read: DB: Prohib/9980 not found in database. -- Executing [s@macro-clr-set:11] Set("SIP/9980-08208aa0", "PROHIB=") in new stack [Sep 10 15:00:17] DEBUG[29884]: app_macro.c:337 _macro_exec: Executed application: Set -- Executing [s@macro-clr-set:12] GotoIf("SIP/9980-08208aa0", "1?100") in new stack -- Goto (macro-clr-set,s,100) [Sep 10 15:00:17] DEBUG[29884]: app_macro.c:337 _macro_exec: Executed application: Gotoif -- Executing [s@macro-clr-set:100] NoOp("SIP/9980-08208aa0", "----") in new stack [Sep 10 15:00:17] DEBUG[29884]: app_macro.c:337 _macro_exec: Executed application: NoOp -- Executing [09981@zebra0:3] Dial("SIP/9980-08208aa0", "SIP/192.168.119.225/09981|600|tT") in new stack Audio is at 192.168.119.251 port 14716 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.119.225:5060: INVITE sip:09981@192.168.119.225 SIP/2.0 Via: SIP/2.0/UDP 192.168.119.251:5060;branch=z9hG4bK772294a0;rport From: "9980" ;tag=as2c19e405 To: Contact: Call-ID: 6febf6532a0b79c2522e35814afbfb10@192.168.119.251 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 10 Sep 2007 13:00:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 266 v=0 o=root 29166 29166 IN IP4 192.168.119.251 s=session c=IN IP4 192.168.119.251 t=0 0 m=audio 14716 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 192.168.119.225/09981 obelix*CLI> <--- SIP read from 192.168.119.225:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.119.251:5060;branch=z9hG4bK772294a0;rport From: "9980" ;tag=as2c19e405 To: ;tag=5074DB18-1C2A Date: Mon, 10 Sep 2007 13:02:20 GMT Call-ID: 6febf6532a0b79c2522e35814afbfb10@192.168.119.251 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 --- obelix*CLI> <--- SIP read from 192.168.119.225:54893 ---> INVITE sip:9981@192.168.119.251:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.119.225:5060;branch=z9hG4bK3033913 Remote-Party-ID: "Wyw" ;party=calling;screen=yes;privacy=off From: "Wyw" ;tag=5074DC14-39 To: Date: Mon, 10 Sep 2007 13:02:20 GMT Call-ID: E6644906-5ED411DC-A6E3E3E2-91BA6B4E@192.168.119.225 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 3865251870-1590956508-3060662285-3989561264 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Timestamp: 1189429340 Contact: Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 234 v=0 o=CiscoSystemsSIP-GW-UserAgent 990 9648 IN IP4 192.168.119.225 s=SIP Call c=IN IP4 192.168.119.225 t=0 0 m=audio 18644 RTP/AVP 18 8 c=IN IP4 192.168.119.225 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 <-------------> --- (21 headers 10 lines) --- Sending to 192.168.119.225 : 5060 (no NAT) Using INVITE request as basis request - E6644906-5ED411DC-A6E3E3E2-91BA6B4E@192.168.119.225 Found user '9980' Found RTP audio format 18 Found RTP audio format 8 [Sep 10 15:00:17] DEBUG[29518]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.119.225:18644 Found description format G729 for ID 18 Found description format PCMA for ID 8 Capabilities: us - 0x108 (alaw|g729), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.119.225:18644 Looking for 9981 in zebra0 (domain 192.168.119.251) <--- Reliably Transmitting (no NAT) to 192.168.119.225:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.119.225:5060;branch=z9hG4bK3033913;received=192.168.119.225 From: "Wyw" ;tag=5074DC14-39 To: ;tag=as06bb56ee Call-ID: E6644906-5ED411DC-A6E3E3E2-91BA6B4E@192.168.119.225 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'E6644906-5ED411DC-A6E3E3E2-91BA6B4E@192.168.119.225' in 32000 ms (Method: INVITE) obelix*CLI> <--- SIP read from 192.168.119.225:5060 ---> ACK sip:9981@192.168.119.251:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.119.225:5060;branch=z9hG4bK3033913 From: "Wyw" ;tag=5074DC14-39 To: ;tag=as06bb56ee Date: Mon, 10 Sep 2007 13:02:20 GMT Call-ID: E6644906-5ED411DC-A6E3E3E2-91BA6B4E@192.168.119.225 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog 'E6644906-5ED411DC-A6E3E3E2-91BA6B4E@192.168.119.225' Method: ACK obelix*CLI> <--- SIP read from 192.168.119.225:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.119.251:5060;branch=z9hG4bK772294a0;rport From: "9980" ;tag=as2c19e405 To: ;tag=5074DB18-1C2A Date: Mon, 10 Sep 2007 13:02:20 GMT Call-ID: 6febf6532a0b79c2522e35814afbfb10@192.168.119.251 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 223 v=0 o=CiscoSystemsSIP-GW-UserAgent 8325 5217 IN IP4 192.168.119.225 s=SIP Call c=IN IP4 192.168.119.225 t=0 0 m=audio 17878 RTP/AVP 18 c=IN IP4 192.168.119.225 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 <-------------> --- (14 headers 10 lines) --- Found RTP audio format 18 [Sep 10 15:00:18] DEBUG[29518]: chan_sip.c:5068 process_sdp: Peer doesn't provide T.38 UDPTL Peer audio RTP is at port 192.168.119.225:17878 Found description format G729 for ID 18 Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.119.225:17878 -- SIP/192.168.119.225-0821a570 is making progress passing it to SIP/9980-08208aa0 Audio is at 192.168.119.251 port 14114 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 192.168.162.2:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.162.2:5060;branch=z9hG4bK4f80994bf9b1bcd5;received=192.168.162.2;rport=5060 From: "9980" ;tag=549b7ff5-704166 To: ;tag=as4cdf403a Call-ID: EA1A-B313-467041664491345C6D4A-801@SipHost CSeq: 801 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 29166 29166 IN IP4 192.168.119.251 s=session c=IN IP4 192.168.119.251 t=0 0 m=audio 14114 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv