JoezSweet
2008-Apr-06 15:42 UTC
[asterisk-users] Help, problems with calls sent from nextone gateway
Hi all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Call audio is fine and all seems well but after 15 to 20 sec the call drops Most of them are dropped while setting up after 5 - 10 sec This fails much more often then it is successful Anyone have a clue on this? Please fine trace below Thanks Joez Trace :- Using INVITE request as basis request - 127191-3416305095-406944 at msx73.mydomain.com Found peer 'enswitch-local' Found RTP audio format 18 Peer audio RTP is at port 82.197.XXX.XXX:20476 Found audio description format G729 for ID 18 Capabilities: us - 0x10c (ulaw|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 82.197.XXX.XXX:20476 Looking for 00556181138037 in from-internal (domain 87.247.224.11) list_route: hop: <sip:87.247.XXX.XXX;lr=on;ftag=3416305095-406953> <--- Transmitting (NAT) to 87.247.224.5:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305095-406953> From: <sip:82.197.XYZ.XYZ:5060>;tag=3416305095-406953 To: 00556181138037 <sip:00556181138037 at 82.197.XYZ.XYZ> Call-ID: 127191-3416305095-406944 at msx73.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:00556181138037 at 87.247.XXX.YYZ> Content-Length: 0 <------------> Audio is at 87.247.XXX.YYZ port 15364 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: INVITE sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef To: <sip:556181138037 at 216.19.ZZZ.ZZZ> Contact: <sip:asterisk at 87.247.XXX.YYZ> Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ CSeq: 102 INVITE User-Agent: Integrics Enswitch Max-Forwards: 70 Date: Fri, 04 Apr 2008 13:31:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=root 2597 2597 IN IP4 87.247.XXX.YYZ s=session c=IN IP4 87.247.XXX.YYZ t=0 0 m=audio 15364 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 556181138037 at voip asterisk2*CLI> <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> SIP/2.0 100 Trying CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6 From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081453123850101510433 Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> Content-Length: 0 <--- SIP read from 87.247.XXX.YYY:5060 ---> CANCEL sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305095-406953> Max-Forwards: 69 To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ> From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305095-406953 Call-ID: 127191-3416305095-406944 at msx73.mydomain.com CSeq: 1 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0 Via: SIP/2.0/UDP 82.197.XYZ.XYZ: 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc Contact: <sip:82.197.XYZ.XYZ:5060> Content-Length: 0 X-Enswitch-Source: 82.197.XYZ.XYZ:5060 X-Enswitch-External: yes Sending to 87.247.XXX.YYY : 5060 (NAT) <--- Reliably Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305095-406953 To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as6ec74197 Call-ID: 127191-3416305095-406944 at msx73.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <--- Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305095-406953> From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305095-406953 To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as6ec74197 Call-ID: 127191-3416305095-406944 at msx73.mydomain.com CSeq: 1 CANCEL User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:00556181138037 at 87.247.XXX.YYZ> Content-Length: 0 <--- SIP read from 87.247.XXX.YYY:5060 ---> ACK sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0 From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305095-406953 Call-ID: 127191-3416305095-406944 at msx73.mydomain.com To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as6ec74197 CSeq: 1 ACK User-Agent: Enswitch SIP proxy Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Scheduling destruction of SIP dialog '04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ ' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: CANCEL sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef To: <sip:556181138037 at 216.19.ZZZ.ZZZ> Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ CSeq: 102 CANCEL User-Agent: Integrics Enswitch Max-Forwards: 70 Content-Length: 0 <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> SIP/2.0 200 OK CSeq: 102 CANCEL Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6 From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081453123850101510433 Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> SIP/2.0 487 Request Terminated CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6 From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081453123850101510433 Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: ACK sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081453123850101510433 Contact: <sip:asterisk at 87.247.XXX.YYZ> Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ CSeq: 102 ACK User-Agent: Integrics Enswitch Max-Forwards: 70 Content-Length: 0 <--- SIP read from 87.247.XXX.YYY:5060 ---> INVITE sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> Max-Forwards: 69 Session-Expires: 3600;Refresher=uac Supported: timer, 100rel To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ> From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 Call-ID: 127193-3416305101-324428 at msx73.mydomain.com CSeq: 1 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0 Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 Contact: <sip:82.197..XYZ.XYZ:5060> Call-Info: <sip:82.197..XYZ.XYZ>;method="NOTIFY;Event=telephone- event;Duration=1000" Content-Type: application/sdp Content-Length: 178 X-Enswitch-Source: 82.197..XYZ.XYZ:5060 X-Enswitch-External: yes v=0 o=msx73 0 0 IN IP4 82.197..XYZ.XYZ s=sip call c=IN IP4 82.197.64.205 t=0 0 m=audio 20500 RTP/AVP 18 a=silenceSupp:on - - - - a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no <-------------> --- (18 headers 9 lines) --- Sending to 87.247.XXX.YYY : 5060 (NAT) Using INVITE request as basis request - 127193-3416305101-324428 at msx73.mydomain.com Found peer 'enswitch-local' Found RTP audio format 18 Peer audio RTP is at port 82.197.64.205:20500 Found audio description format G729 for ID 18 Capabilities: us - 0x10c (ulaw|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 82.197.64.205:20500 Looking for 00556181138037 in from-internal (domain 87.247.XXX.YYZ) list_route: hop: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> asterisk2*CLI> <--- Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0;received=87.247.XXX.YYY Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ> Call-ID: 127193-3416305101-324428 at msx73.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:00556181138037 at 87.247.XXX.YYZ> Content-Length: 0 <------------> Audio is at 87.247.XXX.YYZ port 18712 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: INVITE sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3;rport From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 To: <sip:556181138037 at 216.19.ZZZ.ZZZ> Contact: <sip:asterisk at 87.247.XXX.YYZ> Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ CSeq: 102 INVITE User-Agent: Integrics Enswitch Max-Forwards: 70 Date: Fri, 04 Apr 2008 13:32:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=root 2597 2597 IN IP4 87.247.XXX.YYZ s=session c=IN IP4 87.247.XXX.YYZ t=0 0 m=audio 18712 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 556181138037 at voip asterisk2*CLI> <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> SIP/2.0 100 Trying CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3 From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081459123850695310445 Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk2*CLI> <--- SIP read from 87.247.XXX.YYY:5060 ---> CANCEL sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> Max-Forwards: 69 To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ> From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 Call-ID: 127193-3416305101-324428 at msx73.mydomain.com CSeq: 1 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0 Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 Contact: <sip:82.197..XYZ.XYZ:5060> Content-Length: 0 X-Enswitch-Source: 82.197..XYZ.XYZ:5060 X-Enswitch-External: yes <-------------> --- (14 headers 0 lines) --- Sending to 87.247.XXX.YYY : 5060 (NAT) asterisk2*CLI> <--- Reliably Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0;received=87.247.XXX.YYY Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as109f6054 Call-ID: 127193-3416305101-324428 at msx73.mydomain.com CSeq: 1 INVITE User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0;received=87.247.XXX.YYY Via: SIP/2.0/UDP 82.197..XYZ.XYZ: 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as109f6054 Call-ID: 127193-3416305101-324428 at msx73.mydomain.com CSeq: 1 CANCEL User-Agent: Integrics Enswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:00556181138037 at 87.247.XXX.YYZ> Content-Length: 0 <--- SIP read from 87.247.XXX.YYY:5060 ---> ACK sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0 From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 Call-ID: 127193-3416305101-324428 at msx73.mydomain.com To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as109f6054 CSeq: 1 ACK User-Agent: Enswitch SIP proxy Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Scheduling destruction of SIP dialog '556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ ' in 32000 ms (Method: INVITE) Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: CANCEL sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3;rport From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 To: <sip:556181138037 at 216.19.ZZZ.ZZZ> Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ CSeq: 102 CANCEL User-Agent: Integrics Enswitch Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ ' in 32000 ms (Method: INVITE) == Spawn extension (from-internal, 00556181138037, 1) exited non- zero on 'SIP/5060-088eb4b0' -- Executing [h at from-internal:1] DeadAGI("SIP/5060-088eb4b0", "agi://127.0.0.1/end") in new stack == Spawn extension (to-voip, 00556181138037, 2) exited non-zero on 'Local/00556181138037 at to-voip-f6b9,2' -- Executing [h at to-voip:1] DeadAGI("Local/00556181138037 at to-voip- f6b9,2", "agi://127.0.0.1/end") in new stack -- AGI Script agi://127.0.0.1/end completed, returning 0 asterisk2*CLI> <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> SIP/2.0 200 OK CSeq: 102 CANCEL Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3 From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081459123850695310445 Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> Content-Length: 0 <-------------> --- (8 headers 0 lines) --- asterisk2*CLI> <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> SIP/2.0 487 Request Terminated CSeq: 102 INVITE Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3 From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081459123850695310445 Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> Content-Length: 0
Steve Totaro
2008-Apr-06 16:04 UTC
[asterisk-users] Help, problems with calls sent from nextone gateway
On Sun, Apr 6, 2008 at 11:42 AM, JoezSweet <joezsweet at tiscali.it> wrote:> Hi all, > > I'm having problems with calls dropping after 15 - 20 seconds from a > particular provider. The are using a NexTone gateway. > > Call audio is fine and all seems well but after 15 to 20 sec the call > drops > > Most of them are dropped while setting up after 5 - 10 sec > This fails much more often then it is successful > > Anyone have a clue on this? > Please fine trace below > Thanks > Joez > > Trace :- > > Using INVITE request as basis request - 127191-3416305095-406944 at msx73.mydomain.com > Found peer 'enswitch-local' > Found RTP audio format 18 > Peer audio RTP is at port 82.197.XXX.XXX:20476 > Found audio description format G729 for ID 18 > Capabilities: us - 0x10c (ulaw|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 82.197.XXX.XXX:20476 > Looking for 00556181138037 in from-internal (domain 87.247.224.11) > list_route: hop: <sip:87.247.XXX.XXX;lr=on;ftag=3416305095-406953> > > <--- Transmitting (NAT) to 87.247.224.5:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY > Via: SIP/2.0/UDP 82.197..XYZ.XYZ: > 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc > Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305095-406953> > From: <sip:82.197.XYZ.XYZ:5060>;tag=3416305095-406953 > To: 00556181138037 <sip:00556181138037 at 82.197.XYZ.XYZ> > Call-ID: 127191-3416305095-406944 at msx73.mydomain.com > CSeq: 1 INVITE > User-Agent: Integrics Enswitch > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:00556181138037 at 87.247.XXX.YYZ> > Content-Length: 0 > > > <------------> > Audio is at 87.247.XXX.YYZ port 15364 > Adding codec 0x100 (g729) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: > INVITE sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef > To: <sip:556181138037 at 216.19.ZZZ.ZZZ> > Contact: <sip:asterisk at 87.247.XXX.YYZ> > Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ > CSeq: 102 INVITE > User-Agent: Integrics Enswitch > Max-Forwards: 70 > Date: Fri, 04 Apr 2008 13:31:55 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 263 > > v=0 > o=root 2597 2597 IN IP4 87.247.XXX.YYZ > s=session > c=IN IP4 87.247.XXX.YYZ > t=0 0 > m=audio 15364 RTP/AVP 18 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > --- > -- Called 556181138037 at voip > asterisk2*CLI> > <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> > SIP/2.0 100 Trying > CSeq: 102 INVITE > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6 > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef > Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ > To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081453123850101510433 > Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> > Content-Length: 0 > > > <--- SIP read from 87.247.XXX.YYY:5060 ---> > CANCEL sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 > Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305095-406953> > Max-Forwards: 69 > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ> > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305095-406953 > Call-ID: 127191-3416305095-406944 at msx73.mydomain.com > CSeq: 1 CANCEL > Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, > REFER, SUBSCRIBE, PRACK, UPDATE > Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0 > Via: SIP/2.0/UDP 82.197.XYZ.XYZ: > 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc > Contact: <sip:82.197.XYZ.XYZ:5060> > Content-Length: 0 > X-Enswitch-Source: 82.197.XYZ.XYZ:5060 > X-Enswitch-External: yes > > Sending to 87.247.XXX.YYY : 5060 (NAT) > <--- Reliably Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP > 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY > Via: SIP/2.0/UDP 82.197..XYZ.XYZ: > 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305095-406953 > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as6ec74197 > Call-ID: 127191-3416305095-406944 at msx73.mydomain.com > CSeq: 1 INVITE > User-Agent: Integrics Enswitch > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > > <--- Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0;received=87.247.XXX.YYY > Via: SIP/2.0/UDP 82.197..XYZ.XYZ: > 5060;branch=z9hG4bKac46a19b085b9c507f9c7bffb98e72bc > Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305095-406953> > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305095-406953 > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as6ec74197 > Call-ID: 127191-3416305095-406944 at msx73.mydomain.com > CSeq: 1 CANCEL > User-Agent: Integrics Enswitch > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:00556181138037 at 87.247.XXX.YYZ> > Content-Length: 0 > > > <--- SIP read from 87.247.XXX.YYY:5060 ---> > ACK sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 > Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKc7a3.96bb6b11.0 > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305095-406953 > Call-ID: 127191-3416305095-406944 at msx73.mydomain.com > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as6ec74197 > CSeq: 1 ACK > User-Agent: Enswitch SIP proxy > Content-Length: 0 > > > <-------------> > --- (8 headers 0 lines) --- > Scheduling destruction of SIP dialog '04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ > ' in 32000 ms (Method: INVITE) > Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: > CANCEL sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef > To: <sip:556181138037 at 216.19.ZZZ.ZZZ> > Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ > CSeq: 102 CANCEL > User-Agent: Integrics Enswitch > Max-Forwards: 70 > Content-Length: 0 > > > <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> > SIP/2.0 200 OK > CSeq: 102 CANCEL > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6 > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef > Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ > To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081453123850101510433 > Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> > Content-Length: 0 > > > <-------------> > --- (8 headers 0 lines) --- > <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> > SIP/2.0 487 Request Terminated > CSeq: 102 INVITE > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6 > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef > Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ > To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081453123850101510433 > Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> > Content-Length: 0 > > > <-------------> > --- (8 headers 0 lines) --- > Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: > ACK sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as1f4953ef > To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081453123850101510433 > Contact: <sip:asterisk at 87.247.XXX.YYZ> > Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d at 87.247.XXX.YYZ > CSeq: 102 ACK > User-Agent: Integrics Enswitch > Max-Forwards: 70 > Content-Length: 0 > > > <--- SIP read from 87.247.XXX.YYY:5060 ---> > INVITE sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 > Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> > Max-Forwards: 69 > Session-Expires: 3600;Refresher=uac > Supported: timer, 100rel > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ> > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 > Call-ID: 127193-3416305101-324428 at msx73.mydomain.com > CSeq: 1 INVITE > Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, > REFER, SUBSCRIBE, PRACK, UPDATE > Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0 > Via: SIP/2.0/UDP 82.197..XYZ.XYZ: > 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 > Contact: <sip:82.197..XYZ.XYZ:5060> > Call-Info: <sip:82.197..XYZ.XYZ>;method="NOTIFY;Event=telephone- > event;Duration=1000" > Content-Type: application/sdp > Content-Length: 178 > X-Enswitch-Source: 82.197..XYZ.XYZ:5060 > X-Enswitch-External: yes > > v=0 > o=msx73 0 0 IN IP4 82.197..XYZ.XYZ > s=sip call > c=IN IP4 82.197.64.205 > t=0 0 > m=audio 20500 RTP/AVP 18 > a=silenceSupp:on - - - - > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > > <-------------> > --- (18 headers 9 lines) --- > Sending to 87.247.XXX.YYY : 5060 (NAT) > Using INVITE request as basis request - 127193-3416305101-324428 at msx73.mydomain.com > Found peer 'enswitch-local' > Found RTP audio format 18 > Peer audio RTP is at port 82.197.64.205:20500 > Found audio description format G729 for ID 18 > Capabilities: us - 0x10c (ulaw|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 82.197.64.205:20500 > Looking for 00556181138037 in from-internal (domain 87.247.XXX.YYZ) > list_route: hop: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> > asterisk2*CLI> > <--- Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0;received=87.247.XXX.YYY > Via: SIP/2.0/UDP 82.197..XYZ.XYZ: > 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 > Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ> > Call-ID: 127193-3416305101-324428 at msx73.mydomain.com > CSeq: 1 INVITE > User-Agent: Integrics Enswitch > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:00556181138037 at 87.247.XXX.YYZ> > Content-Length: 0 > > > <------------> > Audio is at 87.247.XXX.YYZ port 18712 > Adding codec 0x100 (g729) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: > INVITE sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3;rport > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 > To: <sip:556181138037 at 216.19.ZZZ.ZZZ> > Contact: <sip:asterisk at 87.247.XXX.YYZ> > Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ > CSeq: 102 INVITE > User-Agent: Integrics Enswitch > Max-Forwards: 70 > Date: Fri, 04 Apr 2008 13:32:00 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 263 > > v=0 > o=root 2597 2597 IN IP4 87.247.XXX.YYZ > s=session > c=IN IP4 87.247.XXX.YYZ > t=0 0 > m=audio 18712 RTP/AVP 18 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > --- > -- Called 556181138037 at voip > asterisk2*CLI> > <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> > SIP/2.0 100 Trying > CSeq: 102 INVITE > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3 > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 > Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ > To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081459123850695310445 > Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> > Content-Length: 0 > > > <-------------> > --- (8 headers 0 lines) --- > asterisk2*CLI> > <--- SIP read from 87.247.XXX.YYY:5060 ---> > CANCEL sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 > Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> > Max-Forwards: 69 > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ> > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 > Call-ID: 127193-3416305101-324428 at msx73.mydomain.com > CSeq: 1 CANCEL > Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, > REFER, SUBSCRIBE, PRACK, UPDATE > Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0 > Via: SIP/2.0/UDP 82.197..XYZ.XYZ: > 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 > Contact: <sip:82.197..XYZ.XYZ:5060> > Content-Length: 0 > X-Enswitch-Source: 82.197..XYZ.XYZ:5060 > X-Enswitch-External: yes > > > <-------------> > --- (14 headers 0 lines) --- > Sending to 87.247.XXX.YYY : 5060 (NAT) > asterisk2*CLI> > <--- Reliably Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP > 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0;received=87.247.XXX.YYY > Via: SIP/2.0/UDP 82.197..XYZ.XYZ: > 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as109f6054 > Call-ID: 127193-3416305101-324428 at msx73.mydomain.com > CSeq: 1 INVITE > User-Agent: Integrics Enswitch > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > > <------------> > > <--- Transmitting (NAT) to 87.247.XXX.YYY:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0;received=87.247.XXX.YYY > Via: SIP/2.0/UDP 82.197..XYZ.XYZ: > 5060;branch=z9hG4bK1c9fe40630b825936f0005d7950b4703 > Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435> > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as109f6054 > Call-ID: 127193-3416305101-324428 at msx73.mydomain.com > CSeq: 1 CANCEL > User-Agent: Integrics Enswitch > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:00556181138037 at 87.247.XXX.YYZ> > Content-Length: 0 > > > <--- SIP read from 87.247.XXX.YYY:5060 ---> > ACK sip:00556181138037 at 87.247.XXX.YYZ:5060 SIP/2.0 > Via: SIP/2.0/UDP 87.247.XXX.YYY;branch=z9hG4bKdbfe.cde834c7.0 > From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435 > Call-ID: 127193-3416305101-324428 at msx73.mydomain.com > To: 00556181138037 <sip:00556181138037 at 82.197..XYZ.XYZ>;tag=as109f6054 > CSeq: 1 ACK > User-Agent: Enswitch SIP proxy > Content-Length: 0 > > > <-------------> > --- (8 headers 0 lines) --- > Scheduling destruction of SIP dialog '556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ > ' in 32000 ms (Method: INVITE) > Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060: > CANCEL sip:556181138037 at 216.19.ZZZ.ZZZ SIP/2.0 > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3;rport > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 > To: <sip:556181138037 at 216.19.ZZZ.ZZZ> > Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ > CSeq: 102 CANCEL > User-Agent: Integrics Enswitch > Max-Forwards: 70 > Content-Length: 0 > > > --- > Scheduling destruction of SIP dialog '556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ > ' in 32000 ms (Method: INVITE) > == Spawn extension (from-internal, 00556181138037, 1) exited non- > zero on 'SIP/5060-088eb4b0' > -- Executing [h at from-internal:1] DeadAGI("SIP/5060-088eb4b0", > "agi://127.0.0.1/end") in new stack > == Spawn extension (to-voip, 00556181138037, 2) exited non-zero on > 'Local/00556181138037 at to-voip-f6b9,2' > -- Executing [h at to-voip:1] DeadAGI("Local/00556181138037 at to-voip- > f6b9,2", "agi://127.0.0.1/end") in new stack > -- AGI Script agi://127.0.0.1/end completed, returning 0 > asterisk2*CLI> > <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> > SIP/2.0 200 OK > CSeq: 102 CANCEL > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3 > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 > Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ > To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081459123850695310445 > Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> > Content-Length: 0 > > > <-------------> > --- (8 headers 0 lines) --- > asterisk2*CLI> > <--- SIP read from 216.19.ZZZ.ZZZ:5060 ---> > SIP/2.0 487 Request Terminated > CSeq: 102 INVITE > Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3 > From: "asterisk" <sip:asterisk at 87.247.XXX.YYZ>;tag=as4b67ecb5 > Call-ID: 556d59211426970524bb236d51144ec4 at 87.247.XXX.YYZ > To: <sip:556181138037 at 216.19.ZZZ.ZZZ>;tag=040431081459123850695310445 > Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp> > Content-Length: 0It appear that your carrier is not answering your call before continuing so the call is timing out. CLI output? Thanks, Steve Totaro