Outbound Dialing Problem using SIP Trunk!

[ZeeTech]

Hi! i am facing probelm while dialing outbound calls however inbounds calls are working on this SIP trunk.

I am using VicidialNOW CE 1.3.

Here is my config.

SIP Carrier.
Registration String register => 001101602:password@SIP-IP
[kaya]
username=001101602
secret=password
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=g7llulaw
host=dynamic
dtmf = rfc2833
qualify=1000
nat=yes
insecure=no


exten => _1XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1XXX.,2,Dial(${SIPTRUNK}/${EXTEN},,tTor)
exten => _1XXX.,3,Hangup

And i can see this SIP Carrier registered
vici*CLI> sip show registry
Host Username Refresh State
188.40.93.13:5060 001101602 105 Registered

I can see following logs while trying dialing outbound.

<-- SIP read from 172.25.86.121:7188:
INVITE sip:13103375200@172.25.90.95 SIP/2.0
To: <sip:13103375200@172.25.90.95>
From: 5000<sip:5000@172.25.90.95>;tag=653df650
Via: SIP/2.0/UDP 172.25.86.121:7188;branch=z9hG4bK-d87543-368392446-1--d87543-;rport
Call-ID: 3721644ee1793d3d
CSeq: 2 INVITE
Contact: <sip:5000@172.25.86.121:7188>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="5000",realm="asterisk",nonce="04ae2e9c",uri="sip:13103375200@172.25.90.95",response="b06d9d01ca62eb3b2687fadc30a7eee0",algorithm=MD5
User-Agent: eyeBeam release 3007n stamp 17816
Content-Length: 276

v=0
o=- 538090553 538090668 IN IP4 172.25.86.121
s=eyeBeam
c=IN IP4 172.25.86.121
t=0 0
m=audio 7198 RTP/AVP 100 6 0 8 3 18 5 101
a=alt:1 1 : CF70136E 00000047 172.25.86.121 7198
a=fmtp:101 0-15
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv

[Apr 13 07:31:13] --- (13 headers 11 lines) ---
[Apr 13 07:31:13] Using INVITE request as basis request - 3721644ee1793d3d
[Apr 13 07:31:13] Sending to 172.25.86.121 : 7188 (NAT)
[Apr 13 07:31:13] Found user '5000'
[Apr 13 07:31:13] Found RTP audio format 100
[Apr 13 07:31:13] Found RTP audio format 6
[Apr 13 07:31:13] Found RTP audio format 0
[Apr 13 07:31:13] Found RTP audio format 8
[Apr 13 07:31:13] Found RTP audio format 3
[Apr 13 07:31:13] Found RTP audio format 18
[Apr 13 07:31:13] Found RTP audio format 5
[Apr 13 07:31:13] Found RTP audio format 101
[Apr 13 07:31:13] Peer audio RTP is at port 172.25.86.121:7198
[Apr 13 07:31:13] Found description format speex
[Apr 13 07:31:13] Found description format telephone-event
[Apr 13 07:31:13] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x32e (gsm|ulaw|alaw|adpcm|g729|speex)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Apr 13 07:31:13] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 13 07:31:13] Looking for 13103375200 in default (domain 172.25.90.95)
[Apr 13 07:31:13] list_route: hop: <sip:5000@172.25.86.121:7188>
[Apr 13 07:31:13] Transmitting (NAT) to 172.25.86.121:7188:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.25.86.121:7188;branch=z9hG4bK-d87543-368392446-1--d87543-;received=172.25.86.121;rport=7188
From: 5000<sip:5000@172.25.90.95>;tag=653df650
To: <sip:13103375200@172.25.90.95>
Call-ID: 3721644ee1793d3d
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:13103375200@172.25.90.95>
Content-Length: 0


---
[Apr 13 07:31:13] -- Executing AGI("SIP/5000-09f1dc48", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 13 07:31:13] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 13 07:31:13] -- Executing Dial("SIP/5000-09f1dc48", "SIP/kaya/+13103375200||tTor") in new stack
[Apr 13 07:31:13] Destroying call '0f667ae618d349f1206ce29b62a4f704@127.0.0.1'
[Apr 13 07:31:13] NOTICE[25327]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
[Apr 13 07:31:13] == Everyone is busy/congested at this time (1:0/0/1)
[Apr 13 07:31:13] -- Executing Hangup("SIP/5000-09f1dc48", "") in new stack
[Apr 13 07:31:13] == Spawn extension (default, 13103375200, 3) exited non-zero on 'SIP/5000-09f1dc48'
[Apr 13 07:31:13] -- Executing DeadAGI("SIP/5000-09f1dc48", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
[Apr 13 07:31:13] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 13 07:31:13] Scheduling destruction of call '3721644ee1793d3d' in 32000 ms
[Apr 13 07:31:13] Reliably Transmitting (NAT) to 172.25.86.121:7188:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.25.86.121:7188;branch=z9hG4bK-d87543-368392446-1--d87543-;received=172.25.86.121;rport=7188
From: 5000<sip:5000@172.25.90.95>;tag=653df650
To: <sip:13103375200@172.25.90.95>;tag=as7267b165
Call-ID: 3721644ee1793d3d
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Please help. FYI Inbound calls are working on this trunk.

[williamconley]

Executing Dial("SIP/5000-09f1dc48", "SIP/kaya/+13103375200||tTor")

Where did that + come from?

[ZeeTech]

No idea

[williamconley]

[Apr 13 07:31:13] -- Executing AGI("SIP/5000-09f1dc48", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 13 07:31:13] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 13 07:31:13] -- Executing Dial("SIP/5000-09f1dc48", "SIP/kaya/+13103375200||tTor") in new stack
[Apr 13 07:31:13] Destroying call '0f667ae618d349f1206ce29b62a4f704@127.0.0.1'
[Apr 13 07:31:13] NOTICE[25327]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)

well, let's start here, i see SIP/kaya, but I don't see the global variable registered to get THAT from the trunk variable listed in the _1 dial plan. What is the global variable set to?

i ask this because it seems that the entire dial plan plays uninterrupted by the actual SIP attempt in those five lines. Like SIP is doing "other stuff" then this runs, the call fails and then SIP is doing other stuff again, but not related to this.

try to isolate the moment of the call with nothing else going on, just to be funny, and get the full path of the call. i would also consider just dialing it from the asterisk command line to keep all the other stuff out of it and see if you come up with something.

[ZeeTech]

Hi William! there was some problem at service provider end; after the changes they made my sip trunk is working with the same settings.

Thanks for your time & considerations.