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[asterisk-users] Help, problems with calls sent from nextone gateway

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stotaro at totarotechn...
Guest

Posted: Sun Apr 06, 2008 11:04 am Post subject: [asterisk-users] Help, problems with calls sent from nextone

On Sun, Apr 6, 2008 at 11:42 AM, JoezSweet <joezsweet at tiscali.it> wrote:

Quote:

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

joezsweet at tiscali.it
Guest

Posted: Sun Apr 06, 2008 12:06 pm Post subject: [asterisk-users] Help, problems with calls sent from nextone

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:

Quote:

On Sun, Apr 6, 2008 at 11:42 AM, JoezSweet <joezsweet at tiscali.it>
wrote:

Quote:

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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