Re: Help,problems with calls sent from nextone gateway

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Help, problems with calls sent from nextone gat...
\ JoezSweet (6 Apr 2008)
. \ Steve Totaro (6 Apr 2008)
. . \ JoezSweet (6 Apr 2008)

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Subject:	Re: Help,problems with calls sent from nextone gateway
Group:	Asterisk-users
From:	JoezSweet
Date:	6 Apr 2008

Hi there,

We get witheld caller cli from client and no cli output, but i dont
think it's the problem.

We had a test with about 200 calls and we got an ACD of about 30 sec,
while from another client with asterisk, for the same route we get
about 3 min ACD.
Beside that we get calls dropped after few sec from invite.

So we're thinking on a compatibility problem between nextone and
asterisk,
or kind of codec stuff we've Digium G729 licensed.

Client configuration is 1 IP for signalling 1 for media and sending G.
729

Any idea on what can be the problem?

Thanks
Giovanni

Il giorno 06/apr/08, alle ore 18:04, Steve Totaro ha scritto:
> On Sun, Apr 6, 2008 at 11:42 AM, JoezSweet <joezsweet>
> 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
>> 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>
>> Call-ID: 127191-3416305095-406944
>> CSeq: 1 INVITE
>> User-Agent: Integrics Enswitch
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Contact: <sip:00556181138037>
>> 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 SIP/2.0
>> Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport
>> From: "asterisk" <sip:asterisk>;tag=as1f4953ef
>> To: <sip:556181138037>
>> Contact: <sip:asterisk>
>> Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d
>> 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@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>;tag=as1f4953ef
>> Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d
>> To: <sip:556181138037>;tag=040431081453123850101510433
>> Contact: <sip:216.19.ZZZ.ZZZ:5060;transport=udp>
>> Content-Length: 0
>>
>>
>> <--- SIP read from 87.247.XXX.YYY:5060 --->
>> CANCEL sip:00556181138037:5060 SIP/2.0
>> Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305095-406953>
>> Max-Forwards: 69
>> To: 00556181138037 <sip:00556181138037>
>> From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305095-406953
>> Call-ID: 127191-3416305095-406944
>> 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>;tag=as6ec74197
>> Call-ID: 127191-3416305095-406944
>> 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>;tag=as6ec74197
>> Call-ID: 127191-3416305095-406944
>> CSeq: 1 CANCEL
>> User-Agent: Integrics Enswitch
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Contact: <sip:00556181138037>
>> Content-Length: 0
>>
>>
>> <--- SIP read from 87.247.XXX.YYY:5060 --->
>> ACK sip:00556181138037:5060 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
>> To: 00556181138037 <sip:
>> 00556181138037>;tag=as6ec74197
>> CSeq: 1 ACK
>> User-Agent: Enswitch SIP proxy
>> Content-Length: 0
>>
>>
>> <------------->
>> --- (8 headers 0 lines) ---
>> Scheduling destruction of SIP dialog '04cd437d3d4ea9d410debb5f3fd2086d
>> ' in 32000 ms (Method: INVITE)
>> Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060:
>> CANCEL sip:556181138037 SIP/2.0
>> Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport
>> From: "asterisk" <sip:asterisk>;tag=as1f4953ef
>> To: <sip:556181138037>
>> Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d
>> 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>;tag=as1f4953ef
>> Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d
>> To: <sip:556181138037>;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>;tag=as1f4953ef
>> Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d
>> To: <sip:556181138037>;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 SIP/2.0
>> Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK5a10cbb6;rport
>> From: "asterisk" <sip:asterisk>;tag=as1f4953ef
>> To: <sip:556181138037>;tag=040431081453123850101510433
>> Contact: <sip:asterisk>
>> Call-ID: 04cd437d3d4ea9d410debb5f3fd2086d
>> CSeq: 102 ACK
>> User-Agent: Integrics Enswitch
>> Max-Forwards: 70
>> Content-Length: 0
>>
>>
>> <--- SIP read from 87.247.XXX.YYY:5060 --->
>> INVITE sip:00556181138037:5060 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>
>> From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435
>> Call-ID: 127193-3416305101-324428
>> 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
>> 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>
>> Call-ID: 127193-3416305101-324428
>> CSeq: 1 INVITE
>> User-Agent: Integrics Enswitch
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Contact: <sip:00556181138037>
>> 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 SIP/2.0
>> Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3;rport
>> From: "asterisk" <sip:asterisk>;tag=as4b67ecb5
>> To: <sip:556181138037>
>> Contact: <sip:asterisk>
>> Call-ID: 556d59211426970524bb236d51144ec4
>> 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@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>;tag=as4b67ecb5
>> Call-ID: 556d59211426970524bb236d51144ec4
>> To: <sip:556181138037>;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:00556181138037:5060 SIP/2.0
>> Record-Route: <sip:87.247.XXX.YYY;lr=on;ftag=3416305101-324435>
>> Max-Forwards: 69
>> To: 00556181138037 <sip:00556181138037>
>> From: <sip:82.197..XYZ.XYZ:5060>;tag=3416305101-324435
>> Call-ID: 127193-3416305101-324428
>> 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>;tag=as109f6054
>> Call-ID: 127193-3416305101-324428
>> 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>;tag=as109f6054
>> Call-ID: 127193-3416305101-324428
>> CSeq: 1 CANCEL
>> User-Agent: Integrics Enswitch
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
>> Supported: replaces
>> Contact: <sip:00556181138037>
>> Content-Length: 0
>>
>>
>> <--- SIP read from 87.247.XXX.YYY:5060 --->
>> ACK sip:00556181138037:5060 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
>> To: 00556181138037 <sip:
>> 00556181138037>;tag=as109f6054
>> CSeq: 1 ACK
>> User-Agent: Enswitch SIP proxy
>> Content-Length: 0
>>
>>
>> <------------->
>> --- (8 headers 0 lines) ---
>> Scheduling destruction of SIP dialog '556d59211426970524bb236d51144ec4
>> ' in 32000 ms (Method: INVITE)
>> Reliably Transmitting (NAT) to 216.19.ZZZ.ZZZ:5060:
>> CANCEL sip:556181138037 SIP/2.0
>> Via: SIP/2.0/UDP 87.247.XXX.YYZ:5060;branch=z9hG4bK112313b3;rport
>> From: "asterisk" <sip:asterisk>;tag=as4b67ecb5
>> To: <sip:556181138037>
>> Call-ID: 556d59211426970524bb236d51144ec4
>> CSeq: 102 CANCEL
>> User-Agent: Integrics Enswitch
>> Max-Forwards: 70
>> Content-Length: 0
>>
>>
>> ---
>> Scheduling destruction of SIP dialog '556d59211426970524bb236d51144ec4
>> ' in 32000 ms (Method: INVITE)
>> == Spawn extension (from-internal, 00556181138037, 1) exited non-
>> zero on 'SIP/5060-088eb4b0'
>> -- Executing [h@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@to-voip-f6b9,2'
>> -- Executing [h@to-voip:1] DeadAGI("Local/00556181138037@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>;tag=as4b67ecb5
>> Call-ID: 556d59211426970524bb236d51144ec4
>> To: <sip:556181138037>;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>;tag=as4b67ecb5
>> Call-ID: 556d59211426970524bb236d51144ec4
>> To: <sip:556181138037>;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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