== Manager 'asteriskDial.exe' logged on from 192.168.16.184 == Using SIP RTP CoS mark 5 Audio is at 192.168.16.8 port 27640 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.16.116:5060: INVITE sip:220@192.168.16.116:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK1d62a8b4;rport Max-Forwards: 70 From: "steve" ;tag=as2f68580d To: Contact: Call-ID: 3916cde07192b44f6cca475662343abe@192.168.16.8 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Date: Wed, 19 Nov 2008 13:59:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 261 v=0 o=root 453129846 453129846 IN IP4 192.168.16.8 s=Asterisk PBX 1.6.0.1 c=IN IP4 192.168.16.8 t=0 0 m=audio 27640 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- localhost*CLI> <--- SIP read from UDP://192.168.16.116:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK1d62a8b4;rport=5060;received=192.168.16.8 From: "steve" ;tag=as2f68580d To: ;tag=314918161 Call-ID: 3916cde07192b44f6cca475662343abe@192.168.16.8 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Contact: Stephen Coles Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- localhost*CLI> <--- SIP read from UDP://192.168.16.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK1d62a8b4;rport=5060;received=192.168.16.8 From: "steve" ;tag=as2f68580d To: ;tag=314918161 Call-ID: 3916cde07192b44f6cca475662343abe@192.168.16.8 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Contact: Stephen Coles Server: Aastra 55i/2.3.1.26 Supported: timer, replaces Content-Type: application/sdp Content-Length: 237 v=0 o=MxSIP 0 0 IN IP4 192.168.16.116 s=SIP Call c=IN IP4 192.168.16.116 t=0 0 m=audio 3000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.16.116:3000 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.16.116:3000 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.16.116, port 5060 Transmitting (no NAT) to 192.168.16.116:5060: ACK sip:220@192.168.16.116:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK795777c0;rport Max-Forwards: 70 From: "steve" ;tag=as2f68580d To: ;tag=314918161 Contact: Call-ID: 3916cde07192b44f6cca475662343abe@192.168.16.8 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 --- > Channel SIP/220-08274360 was answered. -- Executing [222@tapi:1] Macro("SIP/220-08274360", "stdexten,222,SIP/222") in new stack -- Executing [s@macro-stdexten:1] System("SIP/220-08274360", "mysql -u asterisk -h localhost -e "INSERT INTO dialled(extension,number) VALUES('','222')" --password=asterisk asterisk") in new stack == Manager 'asteriskDial.exe' logged off from 192.168.16.184 -- Executing [s@macro-stdexten:2] Dial("SIP/220-08274360", "SIP/222,20") in new stack == Using SIP RTP CoS mark 5 Audio is at 192.168.16.8 port 30728 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.16.162:5060: INVITE sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK138514de;rport Max-Forwards: 70 From: "steve" ;tag=as0e14f0c7 To: Contact: Call-ID: 3f92d34019d7406a5b5f6ce3398243fd@192.168.16.8 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.1 Date: Wed, 19 Nov 2008 13:59:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 261 v=0 o=root 591272716 591272716 IN IP4 192.168.16.8 s=Asterisk PBX 1.6.0.1 c=IN IP4 192.168.16.8 t=0 0 m=audio 30728 RTP/AVP 8 101 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 222 localhost*CLI> <--- SIP read from UDP://192.168.16.162:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK138514de;rport=5060;received=192.168.16.8 From: "steve" ;tag=as0e14f0c7 To: ;tag=3344014224 Call-ID: 3f92d34019d7406a5b5f6ce3398243fd@192.168.16.8 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Contact: Jonathan Holloway Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- -- SIP/222-082ad300 is ringing localhost*CLI> <--- SIP read from UDP://192.168.16.116:5060 ---> BYE sip:asterisk@192.168.16.8 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.116:5060;branch=z9hG4bK4dcfb3ef6559cc4d8.da0ab0c76c4ce243a Max-Forwards: 70 From: ;tag=314918161 To: "steve" ;tag=as2f68580d Call-ID: 3916cde07192b44f6cca475662343abe@192.168.16.8 CSeq: 25939 BYE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Call-Info: ;appearance-index=1 Supported: timer User-Agent: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Sending to 192.168.16.116 : 5060 (no NAT) localhost*CLI> <--- Transmitting (no NAT) to 192.168.16.116:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.116:5060;branch=z9hG4bK4dcfb3ef6559cc4d8.da0ab0c76c4ce243a;received=192.168.16.116 From: ;tag=314918161 To: "steve" ;tag=as2f68580d Call-ID: 3916cde07192b44f6cca475662343abe@192.168.16.8 CSeq: 25939 BYE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3f92d34019d7406a5b5f6ce3398243fd@192.168.16.8' in 32000 ms (Method: INVITE) Reliably Transmitting (no NAT) to 192.168.16.162:5060: CANCEL sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK138514de;rport Max-Forwards: 70 From: "steve" ;tag=as0e14f0c7 To: Call-ID: 3f92d34019d7406a5b5f6ce3398243fd@192.168.16.8 CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 --- Scheduling destruction of SIP dialog '3f92d34019d7406a5b5f6ce3398243fd@192.168.16.8' in 32000 ms (Method: INVITE) == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/220-08274360' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/220-08274360' Really destroying SIP dialog '3916cde07192b44f6cca475662343abe@192.168.16.8' Method: BYE localhost*CLI> <--- SIP read from UDP://192.168.16.162:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK138514de;rport=5060;received=192.168.16.8 From: "steve" ;tag=as0e14f0c7 To: ;tag=3344014224 Call-ID: 3f92d34019d7406a5b5f6ce3398243fd@192.168.16.8 CSeq: 102 CANCEL Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- localhost*CLI> <--- SIP read from UDP://192.168.16.162:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK138514de;rport=5060;received=192.168.16.8 From: "steve" ;tag=as0e14f0c7 To: ;tag=3344014224 Call-ID: 3f92d34019d7406a5b5f6ce3398243fd@192.168.16.8 CSeq: 102 INVITE Server: Aastra 55i/2.3.1.26 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.16.162:5060: ACK sip:222@192.168.16.162:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.16.8:5060;branch=z9hG4bK138514de;rport Max-Forwards: 70 From: "steve" ;tag=as0e14f0c7 To: ;tag=3344014224 Contact: Call-ID: 3f92d34019d7406a5b5f6ce3398243fd@192.168.16.8 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.1 Content-Length: 0 --- Really destroying SIP dialog '3f92d34019d7406a5b5f6ce3398243fd@192.168.16.8' Method: INVITE Really destroying SIP dialog '6cb749cd1b45b268691bf3072e3020c8@127.0.0.1' Method: REGISTER localhost*CLI>