[asterisk-users] Help, problems with calls sent from nextone gateway
Steve Totaro
stotaro at totarotechnologies.com
Sun Apr 6 11:04:13 CDT 2008
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: 0
It appear that your carrier is not answering your call before
continuing so the call is timing out. CLI output?
Thanks,
Steve Totaro
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