Here somewhere int the world a phone 987654 is calling 123456. In the following log : the provider trunk have address 1.1.1.251 the provider RTP gateways have address 1.1.1.x. the asterisk address is 2.2.2.2 the custormer PABX have address 3.3.3.251 The interesting part is that the PABX returns an RFC3398 compliant result : <--- SIP read from UDP:3.3.3.251:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK4acf49d4;rport=5060;received=2.2.2.2 From: "987654" ;tag=as2eff6979 To: ;tag=3169669864 Call-ID: 27f7d0e862a52b933eeaf4c357b041bd@2.2.2.2:5060 CSeq: 102 INVITE Server: Patton SN4638 5BIS 00A0BA06BAEF R6.1 2012-05-09 H323 SIP BRI M5T SIP Stack/4.0.30.30 Content-Length: 0 and this is forwarded by asterisk to the provider like this : <--- Reliably Transmitting (NAT) to 1.1.1.104:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 1.1.1.104;branch=z9hG4bK1166.4a4d19f6.1;received=1.1.1.104;rport=5060 Via: SIP/2.0/UDP 1.1.1.116:5060;rport=5060;branch=z9hG4bK-77ef-1346673299-2623-361 From: "987654";tag=95ffcd055e0f78f7d5d397020e89288d9808247e To: ;tag=as4447fef1 Call-ID: 2c00-4b2-832012115459-ACKBAR-0-1.1.1.116 CSeq: 1 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Reason: Q.850;cause=28 X-Asterisk-HangupCause: Invalid number format X-Asterisk-HangupCauseCode: 28 Content-Length: 0 The other interesting part is that the "484 Address Incomplete" generates a DIALSTATUS=CHANUNAVAIL. The full log : <--- SIP read from UDP:1.1.1.104:5060 ---> INVITE sip:123456@2.2.2.2 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 1.1.1.104;branch=z9hG4bK1166.4a4d19f6.1 Via: SIP/2.0/UDP 1.1.1.116:5060;rport=5060;branch=z9hG4bK-77ef-1346673299-2623-361 Call-ID: 2c00-4b2-832012115459-ACKBAR-0-1.1.1.116 CSeq: 1 INVITE Max-Forwards: 16 To: From: "987654";tag=95ffcd055e0f78f7d5d397020e89288d9808247e User-Agent: Dialogic-SIP/10.5.3.333 ACKBAR 0 Timestamp: 09032012115459 P-Asserted-Identity: "987654" Contact: Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Supported: path, replaces, timer, tdialog Session-Expires: 1800 Expires: 300 Organization: Dialogic Content-Type: application/sdp Content-Length: 302 v=0 o=Dialogic_SDP 5866848 0 IN IP4 1.1.1.116 s=Dialogic-SIP c=IN IP4 1.1.1.123 t=0 0 m=audio 8712 RTP/AVP 8 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - <-------------> --- (20 headers 13 lines) --- Sending to 1.1.1.104:5060 (NAT) Using INVITE request as basis request - 2c00-4b2-832012115459-ACKBAR-0-1.1.1.116 Found peer 'legos-eliance' for '987654' from 1.1.1.104:5060 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=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 1.1.1.123:8712 Looking for 123456 in legos-in (domain 2.2.2.2) list_route: hop: <--- Transmitting (NAT) to 1.1.1.104:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.104;branch=z9hG4bK1166.4a4d19f6.1;received=1.1.1.104;rport=5060 Via: SIP/2.0/UDP 1.1.1.116:5060;rport=5060;branch=z9hG4bK-77ef-1346673299-2623-361 Record-Route: From: "987654";tag=95ffcd055e0f78f7d5d397020e89288d9808247e To: Call-ID: 2c00-4b2-832012115459-ACKBAR-0-1.1.1.116 CSeq: 1 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <------------> == 2012-09-03-13:55:04.000 equation-iad1 calling 123456 callerid num=987654 name=987654 num-pres=allowed_not_screened name-pres=allowed_not_screened == 2012-09-03-13:55:04.000 equation-iad1 There are 1 calls == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 12414 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 3.3.3.251:5060: INVITE sip:123456@3.3.3.251 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK4acf49d4;rport Max-Forwards: 16 From: "987654" ;tag=as2eff6979 To: Contact: Call-ID: 27f7d0e862a52b933eeaf4c357b041bd@2.2.2.2:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Mon, 03 Sep 2012 11:55:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "987654" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 281 v=0 o=root 1837498627 1837498627 IN IP4 2.2.2.2 s=Asterisk PBX c=IN IP4 2.2.2.2 t=0 0 m=audio 12414 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #1 (NAT) to 3.3.3.251:5060: INVITE sip:123456@3.3.3.251 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK4acf49d4;rport Max-Forwards: 16 From: "987654" ;tag=as2eff6979 To: Contact: Call-ID: 27f7d0e862a52b933eeaf4c357b041bd@2.2.2.2:5060 CSeq: 102 INVITE User-Agent: Asterisk Date: Mon, 03 Sep 2012 11:55:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "987654" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 281 v=0 o=root 1837498627 1837498627 IN IP4 2.2.2.2 s=Asterisk PBX c=IN IP4 2.2.2.2 t=0 0 m=audio 12414 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:3.3.3.251:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK4acf49d4;rport=5060;received=2.2.2.2 From: "987654" ;tag=as2eff6979 To: ;tag=3169669864 Call-ID: 27f7d0e862a52b933eeaf4c357b041bd@2.2.2.2:5060 CSeq: 102 INVITE Contact: Server: Patton SN4638 5BIS 00A0BA06BAEF R6.1 2012-05-09 H323 SIP BRI M5T SIP Stack/4.0.30.30 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- list_route: hop: <--- Transmitting (NAT) to 1.1.1.104:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 1.1.1.104;branch=z9hG4bK1166.4a4d19f6.1;received=1.1.1.104;rport=5060 Via: SIP/2.0/UDP 1.1.1.116:5060;rport=5060;branch=z9hG4bK-77ef-1346673299-2623-361 Record-Route: From: "987654";tag=95ffcd055e0f78f7d5d397020e89288d9808247e To: ;tag=as4447fef1 Call-ID: 2c00-4b2-832012115459-ACKBAR-0-1.1.1.116 CSeq: 1 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <--- SIP read from UDP:3.3.3.251:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK4acf49d4;rport=5060;received=2.2.2.2 From: "987654" ;tag=as2eff6979 To: ;tag=3169669864 Call-ID: 27f7d0e862a52b933eeaf4c357b041bd@2.2.2.2:5060 CSeq: 102 INVITE Server: Patton SN4638 5BIS 00A0BA06BAEF R6.1 2012-05-09 H323 SIP BRI M5T SIP Stack/4.0.30.30 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 3.3.3.251:5060 Transmitting (NAT) to 3.3.3.251:5060: ACK sip:123456@3.3.3.251:5060 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK4acf49d4;rport Max-Forwards: 16 From: "987654" ;tag=as2eff6979 To: ;tag=3169669864 Contact: Call-ID: 27f7d0e862a52b933eeaf4c357b041bd@2.2.2.2:5060 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- == Everyone is busy/congested at this time (1:0/0/1) == 2012-09-03-13:55:11.000 equation-iad1 DIALSTATUS = CHANUNAVAIL == 2012-09-03-13:55:11.000 equation-iad1 HANGUPCAUSE = 28 <--- Reliably Transmitting (NAT) to 1.1.1.104:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 1.1.1.104;branch=z9hG4bK1166.4a4d19f6.1;received=1.1.1.104;rport=5060 Via: SIP/2.0/UDP 1.1.1.116:5060;rport=5060;branch=z9hG4bK-77ef-1346673299-2623-361 From: "987654";tag=95ffcd055e0f78f7d5d397020e89288d9808247e To: ;tag=as4447fef1 Call-ID: 2c00-4b2-832012115459-ACKBAR-0-1.1.1.116 CSeq: 1 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Reason: Q.850;cause=28 X-Asterisk-HangupCause: Invalid number format X-Asterisk-HangupCauseCode: 28 Content-Length: 0 <------------> Really destroying SIP dialog '27f7d0e862a52b933eeaf4c357b041bd@2.2.2.2:5060' Method: INVITE <--- SIP read from UDP:1.1.1.104:5060 ---> ACK sip:123456@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.104;branch=z9hG4bK1166.4a4d19f6.1 From: "987654";tag=95ffcd055e0f78f7d5d397020e89288d9808247e Call-ID: 2c00-4b2-832012115459-ACKBAR-0-1.1.1.116 To: ;tag=as4447fef1 CSeq: 1 ACK User-Agent: Bornsip BT (1.2) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '2c00-4b2-832012115459-ACKBAR-0-1.1.1.116' Method: ACK