Audio is at 14422 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100002 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.4.150:1161: INVITE sip:10@192.168.4.150:1161;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.4.40:5060;branch=z9hG4bK25c21ff6;rport Max-Forwards: 70 From: "dongle0" ;tag=as78568dda To: Contact: Call-ID: 7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 10.12.1 Date: Fri, 16 Aug 2013 10:25:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 308 v=0 o=root 975828700 975828700 IN IP4 192.168.4.40 s=Asterisk PBX 10.12.1 c=IN IP4 192.168.4.40 t=0 0 m=audio 14422 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from TCP:192.168.4.150:1161 ---> SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.4.40:5060;branch=z9hG4bK25c21ff6;rport=5060 From: "dongle0" ;tag=as78568dda To: Call-ID: 7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060 CSeq: 102 INVITE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from TCP:192.168.4.150:1161 ---> SIP/2.0 180 Ringing Via: SIP/2.0/TCP 192.168.4.40:5060;branch=z9hG4bK25c21ff6;rport=5060 From: "dongle0" ;tag=as78568dda To: ;tag=111118581 Call-ID: 7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060 CSeq: 102 INVITE Contact: User-Agent: DLINK DPH-150S FRU2.2.162.67 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- list_route: hop: -- SIP/10-00000017 is ringing -- SIP/11-00000016 is ringing -- Stopped music on hold on Dongle/dongle0-010000000b Scheduling destruction of SIP dialog '7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 192.168.4.150:1161: CANCEL sip:10@192.168.4.150:1161;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.4.40:5060;branch=z9hG4bK25c21ff6;rport Max-Forwards: 70 From: "dongle0" ;tag=as78568dda To: Call-ID: 7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060 CSeq: 102 CANCEL User-Agent: Asterisk PBX 10.12.1 Content-Length: 0 --- Scheduling destruction of SIP dialog '7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060' in 32000 ms (Method: INVITE) == Spawn extension (datacard-incoming, +79114000000, 2) exited non-zero on 'Dongle/dongle0-010000000b' == MixMonitor close filestream (mixed) == Executing [/usr/bin/nice -n 19 /usr/bin/lame -b 16 --silent "/tmp/asterisk/wav/1376648707.33.wav" "/var/www/asterisk/mp3/1376648707.33.mp3" && rm -f "/tmp/asterisk/wav/1376648707.33.wav" && chmod 755 "/var/www/asterisk/mp3/1376648707.33.mp3"] <--- SIP read from TCP:192.168.4.150:1161 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.4.40:5060;branch=z9hG4bK25c21ff6;rport=5060 From: "dongle0" ;tag=as78568dda To: ;tag=111118581 Call-ID: 7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060 CSeq: 102 CANCEL User-Agent: DLINK DPH-150S FRU2.2.162.67 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from TCP:192.168.4.150:1161 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/TCP 192.168.4.40:5060;branch=z9hG4bK25c21ff6;rport=5060 From: "dongle0" ;tag=as78568dda To: ;tag=111118581 Call-ID: 7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060 CSeq: 102 INVITE User-Agent: DLINK DPH-150S FRU2.2.162.67 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (NAT) to 192.168.4.150:1161: ACK sip:10@192.168.4.150:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 192.168.4.40:5060;branch=z9hG4bK25c21ff6;rport Max-Forwards: 70 From: "dongle0" ;tag=as78568dda To: ;tag=111118581 Contact: Call-ID: 7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 10.12.1 Content-Length: 0 --- Scheduling destruction of SIP dialog '7bcd9b461939a38a04aae16d76df2e54@192.168.4.40:5060' in 32000 ms (Method: INVITE) == End MixMonitor Recording Dongle/dongle0-010000000b <--- SIP read from TCP:192.168.4.150:1161 ---> OPTIONS sip:192.168.4.40:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.4.150:1161;branch=z9hG4bK2307176041699511747 From: "10" ;tag=481528454 To: Call-ID: 27505727515424-5904684132712@192.168.4.150 CSeq: 1 OPTIONS Max-Forwards: 70 User-Agent: DLINK DPH-150S FRU2.2.162.67 Accept: application/sdp Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Looking for s in default (domain 192.168.4.40) <--- Transmitting (NAT) to 192.168.4.150:1161 ---> SIP/2.0 404 Not Found Via: SIP/2.0/TCP 192.168.4.150:1161;branch=z9hG4bK2307176041699511747;received=192.168.4.150;rport=1161 From: "10" ;tag=481528454 To: ;tag=as125caa93 Call-ID: 27505727515424-5904684132712@192.168.4.150 CSeq: 1 OPTIONS Server: Asterisk PBX 10.12.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '27505727515424-5904684132712@192.168.4.150' in 32000 ms (Method: OPTIONS) debian*CLI> sip set debug off SIP Debugging Disabled