*CLI> sip set debug on SIP Debugging enabled *CLI> <--- SIP read from UDP:127.0.0.1:5062 ---> INVITE sip:8690@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKobts2861b16636075e7730 From: IMSI466990004244439 ;tag=larfbjaqwuutxjag To: Call-ID: 414093030@127.0.0.1 CSeq: 388 INVITE Contact: ;expires=3600 Content-Type: application/sdp User-Agent: OpenBTS P2.8TRUNK Build Date Dec 20 2012 Max-Forwards: 5 P-Access-Network-Info: 3GPP-GERAN; cgi-3gpp=4669703e8000a P-Preferred-Identity: Content-Length: 135 v=0 o=IMSI466990004244439 0 0 IN IP4 127.0.0.1 s=Talk Time t=0 0 m=audio 16500 RTP/AVP 3 c=IN IP4 127.0.0.1 a=rtpmap:3 GSM/8000 <-------------> --- (13 headers 7 lines) --- Sending to 127.0.0.1:5062 (NAT) Using INVITE request as basis request - 414093030@127.0.0.1 Found peer 'IMSI466990004244439' for 'IMSI466990004244439' from 127.0.0.1:5062 == Using SIP RTP CoS mark 5 Found RTP audio format 3 Found audio description format GSM for ID 3 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 127.0.0.1:16500 Looking for 8690 in phones (domain 127.0.0.1) list_route: hop: <--- Transmitting (NAT) to 127.0.0.1:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKobts2861b16636075e7730;received=127.0.0.1;rport=5062 From: IMSI466990004244439 ;tag=larfbjaqwuutxjag To: Call-ID: 414093030@127.0.0.1 CSeq: 388 INVITE Server: Asterisk PBX 1.8.11-cert9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [8690@phones:1] Dial("SIP/IMSI466990004244439-00000014", "SIP/IMSI466974104638690") in new stack Really destroying SIP dialog '3862c8d23be16ce36e564c3251cbc10c@127.0.1.1:5060' Method: INVITE [Dec 21 00:05:39] WARNING[2838]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/IMSI466990004244439-00000014' status is 'CHANUNAVAIL' <--- Reliably Transmitting (NAT) to 127.0.0.1:5062 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKobts2861b16636075e7730;received=127.0.0.1;rport=5062 From: IMSI466990004244439 ;tag=larfbjaqwuutxjag To: ;tag=as213c65f3 Call-ID: 414093030@127.0.0.1 CSeq: 388 INVITE Server: Asterisk PBX 1.8.11-cert9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 20 Content-Length: 0 <------------> <--- SIP read from UDP:127.0.0.1:5062 ---> ACK sip:8690@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKobts2861b16636075e7730 From: IMSI466990004244439 ;tag=larfbjaqwuutxjag To: ;tag=as213c65f3 Call-ID: 414093030@127.0.0.1 CSeq: 388 ACK User-Agent: OpenBTS P2.8TRUNK Build Date Dec 20 2012 Max-Forwards: 5 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '414093030@127.0.0.1' Method: ACK <--- SIP read from UDP:127.0.0.1:5062 ---> CANCEL sip:IMSI466990004244439@127.0.0.1:5062 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKobts2861b16636075e7730 From: IMSI466990004244439 ;tag=larfbjaqwuutxjag To: Call-ID: 414093030@127.0.0.1 CSeq: 388 CANCEL Contact: ;expires=3600 User-Agent: OpenBTS P2.8TRUNK Build Date Dec 20 2012 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- Transmitting (NAT) to 127.0.0.1:5062 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bKobts2861b16636075e7730;received=127.0.0.1;rport=5062 From: IMSI466990004244439 ;tag=larfbjaqwuutxjag To: ;tag=as5bb4351f Call-ID: 414093030@127.0.0.1 CSeq: 388 CANCEL Server: Asterisk PBX 1.8.11-cert9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> *CLI> *CLI>