In-group cant call SIP

[jselvin]

Hello, I'm trying to set up an in-group for our customer services. I'm following Tutorial E in the managers manual.

When I try dialing 0855118000- I get no sound at all, and after about 15 seconds, it signals busy in my phone (that I use to call in).

Does anyone know what could be wrong?

Kind regards
Jimmy Selvin

Asterisk 1.2.30.4
VERSION: 2.2.0-236
BUILD: 100413-2328


SIP:
Carrier ->
[cellip]
type=peer
username=46855118000
fromuser=46855118000
host=sip.mysecretary.net
fromdomain=sip.mysecretary.net
insecure=very
canreinvite=no
context=trunkinbound

Dialplan
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(SIP/cellip/${EXTEN},55,tTo)
exten => _X.,3,Hangup
______________

AsteriskCLI ->

Connected to Asterisk 1.2.30.4 currently running on vicibox (pid = 6482)
Verbosity is at least 21
[Apr 25 00:09:16] -- Remote UNIX connection
[Apr 25 00:09:27] -- Executing AGI("SIP/46855118000-081e7d28", "agi-DID_route.agi") in new stack
[Apr 25 00:09:27] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Apr 25 00:09:27] -- AGI Script agi-DID_route.agi completed, returning 0
[Apr 25 00:09:27] -- Executing Wait("SIP/46855118000-081e7d28","2") in new stack
[Apr 25 00:09:29] -- Executing Answer("SIP/46855118000-081e7d28", "") in new stack
[Apr 25 00:09:29] -- Executing Playback("SIP/46855118000-081e7d28", "ss-no service") in new stack
[Apr 25 00:09:29] -- Playing 'ss-noservice' (language 'en')
[Apr 25 00:09:34] -- Executing Playback("SIP/46855118000-081e7d28", "vm-goodbye") in new stack
[Apr 25 00:09:34] -- Playing 'vm-goodbye' (language 'en')
[Apr 25 00:09:35] -- Executing Hangup("SIP/46855118000-081e7d28", "") in new stack
[Apr 25 00:09:35] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/46855118000-081e7d28'
[Apr 25 00:09:35] -- Executing DeadAGI("SIP/46855118000-081e7d28", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 25 00:09:35] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----16--------------- completed, returning 0

[mflorell]

Please post agiout logfile output for this call

[jselvin]

Hi, still doesn't work, and I've tried a lot of different things..

If I haven't missed anything obvious, It could be because I'm on a VPN connection? I have no problems dialing the number when I'm connected through the providers voip Server... Could this be related to the provider?

Hope you have some ideas.

Kind regards
Jimmy Selvin

2010-04-26 01:11:37|agi-DID_route.agi||INSERT INTO vicidial_did_log SET uniqueid='1272237097.108',channel='SIP/46855118010-081fbd20',server_ip='10.27.98$
2010-04-26 01:11:37|agi-DID_route.agi|-- DID LOG : |1|INSERT INTO vicidial_did_log SET uniqueid='1272237097.108',channel='SIP/46855118010-081fbd20',s$
2010-04-26 01:11:37|agi-DID_route.agi|-- CALL LOG : |1|INSERT INTO call_log SET uniqueid='1272237097.108', channel='SIP/46855118010-081fbd20', channe$
2010-04-26 01:11:37|agi-DID_route.agi|exiting the DID app, transferring call to 9998811112 @ default


CLI:
[Apr 26 01:13:11] --- (0 headers 1 lines) ---
[Apr 26 01:13:39] -- Executing AGI("SIP/46855118010-081fbd20", "agi-DID_route.agi") in new stack
[Apr 26 01:13:39] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Apr 26 01:13:39] -- AGI Script agi-DID_route.agi completed, returning 0
[Apr 26 01:13:39] -- Executing Wait("SIP/46855118010-081fbd20", "2") in new stack
[Apr 26 01:13:41]
<-- SIP read from 10.27.97.4:40779:



[Apr 26 01:13:41] --- (0 headers 1 lines) ---
[Apr 26 01:13:42] -- Executing Answer("SIP/46855118010-081fbd20", "") in new stack
[Apr 26 01:13:42] -- Executing Playback("SIP/46855118010-081fbd20", "ss-noservice") in new stack
[Apr 26 01:13:42] -- Playing 'ss-noservice' (language 'en')
[Apr 26 01:13:50] -- Executing Playback("SIP/46855118010-081fbd20", "vm-goodbye") in new stack
[Apr 26 01:13:50] -- Playing 'vm-goodbye' (language 'en')
[Apr 26 01:13:50] -- Executing Hangup("SIP/46855118010-081fbd20", "") in new stack
[Apr 26 01:13:50] == Spawn extension (default, 9998811112, 5) exited non-zero on 'SIP/46855118010-081fbd20'
[Apr 26 01:13:50] -- Executing DeadAGI("SIP/46855118010-081fbd20", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 26 01:13:50] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 26 01:14:07] == Parsing '/etc/asterisk/manager.conf': [Apr 26 01:14:07] Found
[Apr 26 01:14:07] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 26 01:14:07] == Parsing '/etc/asterisk/manager.conf': [Apr 26 01:14:07] Found
[Apr 26 01:14:07] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 26 01:14:07] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 26 01:14:11]
<-- SIP read from 10.27.97.4:40779:





[Apr 26 01:16:41] --- (0 headers 1 lines) ---
[Apr 26 01:16:48]
<-- SIP read from 10.27.97.4:40779:
REGISTER sip:10.27.98.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:40779;branch=z9hG4bK-d8754z-6b787d1a6834d50c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc150@10.27.97.4:40779;rinstance=b9c987cff9e2db26>
To: "cc150"<sip:cc150@10.27.98.3>
From: "cc150"<sip:cc150@10.27.98.3>;tag=340c3c3c
Call-ID: NTUwZmMwZDk1ZDdhYzVhZmVhNzljMDMxNWQ3ZWMzY2M.
CSeq: 10 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="cc150",realm="asterisk",nonce="237dfcb7",uri="sip:10.27.98.3",response="1a255e7f528e10caf410bf3c11c302c8",algorithm=MD5
Content-Length: 0


[Apr 26 01:16:48] --- (13 headers 0 lines) ---
[Apr 26 01:16:48] Using latest REGISTER request as basis request
[Apr 26 01:16:48] Sending to 192.168.0.13 : 40779 (NAT)
[Apr 26 01:16:48] Transmitting (NAT) to 10.27.97.4:40779:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.13:40779;branch=z9hG4bK-d8754z-6b787d1a6834d50c-1---d8754z-;received=10.27.97.4;rport=40779
From: "cc150"<sip:cc150@10.27.98.3>;tag=340c3c3c
To: "cc150"<sip:cc150@10.27.98.3>
Call-ID: NTUwZmMwZDk1ZDdhYzVhZmVhNzljMDMxNWQ3ZWMzY2M.
CSeq: 10 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:cc150@10.27.98.3>
Content-Length: 0


---
[Apr 26 01:16:48] Transmitting (NAT) to 10.27.97.4:40779:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.13:40779;branch=z9hG4bK-d8754z-6b787d1a6834d50c-1---d8754z-;received=10.27.97.4;rport=40779
From: "cc150"<sip:cc150@10.27.98.3>;tag=340c3c3c
To: "cc150"<sip:cc150@10.27.98.3>;tag=as758f04bb
Call-ID: NTUwZmMwZDk1ZDdhYzVhZmVhNzljMDMxNWQ3ZWMzY2M.
CSeq: 10 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e7fc870"
Content-Length: 0


---
[Apr 26 01:16:48] Scheduling destruction of call 'NTUwZmMwZDk1ZDdhYzVhZmVhNzljMDMxNWQ3ZWMzY2M.' in 15000 ms
[Apr 26 01:16:48]
<-- SIP read from 10.27.97.4:40779:
REGISTER sip:10.27.98.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.13:40779;branch=z9hG4bK-d8754z-f8249e46851a1c22-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc150@10.27.97.4:40779;rinstance=b9c987cff9e2db26>
To: "cc150"<sip:cc150@10.27.98.3>
From: "cc150"<sip:cc150@10.27.98.3>;tag=340c3c3c
Call-ID: NTUwZmMwZDk1ZDdhYzVhZmVhNzljMDMxNWQ3ZWMzY2M.
CSeq: 11 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Authorization: Digest username="cc150",realm="asterisk",nonce="0e7fc870",uri="sip:10.27.98.3",response="f5a4f2319fc8a337d9184eda63d03aec",algorithm=MD5
Content-Length: 0


[Apr 26 01:16:48] --- (13 headers 0 lines) ---
[Apr 26 01:16:48] Using latest REGISTER request as basis request
[Apr 26 01:16:48] Sending to 192.168.0.13 : 40779 (NAT)
[Apr 26 01:16:48] Transmitting (NAT) to 10.27.97.4:40779:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.13:40779;branch=z9hG4bK-d8754z-f8249e46851a1c22-1---d8754z-;received=10.27.97.4;rport=40779
From: "cc150"<sip:cc150@10.27.98.3>;tag=340c3c3c
To: "cc150"<sip:cc150@10.27.98.3>
Call-ID: NTUwZmMwZDk1ZDdhYzVhZmVhNzljMDMxNWQ3ZWMzY2M.
CSeq: 11 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:cc150@10.27.98.3>
Content-Length: 0