Not a valid extension

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Not a valid extension

Postby john_usc » Fri Jul 17, 2009 1:13 am

I am using VERSION: 2.0.5-173 BUILD: 90320-0424 CentOS 5

Here is my problem

I can make the calls manually using softphone, which means that my provider account is setup ok. However when agent logs in , instead of hearing you are the only person in this conference, it plays this message

Its an invalid extension

This is what I get on CLI
== CDR updated on SIP/202-0863c8c0
-- Executing Goto("SIP/202-0863c8c0", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("SIP/202-0863c8c0", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Executing Hangup("SIP/202-0863c8c0", "") in new stack
== Spawn extension (default, #, 2) exited non-zero on 'SIP/202-0863c8c0'
-- Executing DeadAGI("SIP/202-0863c8c0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-----

here is my extension.conf settings for out bound calls

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@${TRUNKSIP1},55,o)
;exten => _91NXXNXXXXXX,3,Dial(sip/${EXTEN:1}@${TRUNKSIP2},55,o)
exten => _91NXXNXXXXXX,3,Hangup
exten => 201,1,Dial(SIP/201)
exten => 201,2,Voicemail,u201
exten => 202,1,Dial(SIP/202)
exten => 202,2,Voicemail,u202

Not sure what could be causing this

Thanks
john_usc
 
Posts: 167
Joined: Sat Nov 08, 2008 9:59 pm

also get this on the browser

Postby john_usc » Fri Jul 17, 2009 1:18 am

Sorry, there are no available sessions
john_usc
 
Posts: 167
Joined: Sat Nov 08, 2008 9:59 pm

i tired this

Postby john_usc » Fri Jul 17, 2009 1:27 am

I tried adding this on top of extensions.conf

[default]

; BE SURE TO CHANGE THIS LINE FOR YOUR IP ADDRESS!
exten => _192*168*002*015*.,1,Goto(default,${EXTEN:16},1)
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)

and now it doesnt play message but still agent cant login and I get this on CLI

== Starting SIP/202-0863c8c0 at default,,1 failed so falling back to exten 's'
== Starting SIP/202-0863c8c0 at default,s,1 still failed so falling back to context 'default'
Jul 16 22:28:18 WARNING[23010]: pbx.c:2377 __ast_pbx_run: Channel 'SIP/202-0863c8c0' sent into invalid extension 's' in context 'default', but no invalid handler


thanks
john_usc
 
Posts: 167
Joined: Sat Nov 08, 2008 9:59 pm

Re: also get this on the browser

Postby webgurru » Fri Jul 17, 2009 3:10 am

Are you sure your SIP acount is properly configured. Check output of this command on asterisk CLI

Code: Select all
sip show registry


john_usc wrote:Sorry, there are no available sessions
webgurru
 
Posts: 147
Joined: Thu May 07, 2009 11:10 am
Location: United Kingdom

Postby john_usc » Fri Jul 17, 2009 9:46 pm

I am testing out the newer version and it was suppose to be easy to add carriers but I am having a hack of the time. I have the older version working fine. I am just not sure why this newer version isnt working for me. I went through managers manual and I first made a Phone 201 and then entered the carrier exactly as said in managers manual with some obvious exceptions. here is what I did

Carrier ID: SIPVOICE
Carrier Name: VOICEPULSE
Registration String:
register => user:password@jfk-primary.voicepulse.com
Template ID: None
Account Entry:
[voicepulse-primary]
type=peer
host=jfk-primary.voicepulse.com
username=user
secret=password
qualify=yes
allow=all
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes
context=vicidial-auto

Dialplan Entry:
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(user:pass@jfk-primary.voicepulse.com/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup



I didnt use context=trunckinbound as I am dialing outbound campaigs. I tried context=default and context=vicidial-auto but nothing is working

When I go to softphone (x-lite) I dial a number like 918056453353 (in format of 91NxxNxxxxxx) it gives me an announcment that "The Person you are calling is unavailable)

and CLI gives this

-- Executing AGI("SIP/201-b782fee8", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/201-b782fee8", "/17203334762||To") in new stack
Jul 17 18:43:49 WARNING[18677]: channel.c:2621 ast_request: No channel type registered for ''
Jul 17 18:43:49 NOTICE[18677]: app_dial.c:1076 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/201-b782fee8", "") in new stack
== Spawn extension (default, 917203334762, 3) exited non-zero on 'SIP/201-b782fee8'
-- Executing DeadAGI("SIP/201-b782fee8", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----66-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


WHEN I DO : SIP SHOW PEERS

I can see it in cli as registered fine

201/201 192.168.2.11 D N 61890 OK (116 ms)
voicepulse-primary/user 64.61.93.190 N 5060 OK (65 ms)

WHEN I DO : SIP SHOW REGISTRY

I can see the provider is also listed

Host Username Refresh State
jfk-primary.voicepulse.com:506 user 105 Registered


Its driving me crazy. Please help!
Thanks
john_usc
 
Posts: 167
Joined: Sat Nov 08, 2008 9:59 pm

Postby john_usc » Sat Jul 18, 2009 12:13 am

ok I fixed that problem and this is how I did it.

I reinstall vicidial and then I didnt follow the managers manual (paid version). I actually used this free guide here
http://taguinod.net/vnow/vnow-getting-started-draft.pdf

I followed this guide. I used context=inbound as said in the guide and it worked. I still want to find out how the context work.
Hope this will help someone.
john_usc
 
Posts: 167
Joined: Sat Nov 08, 2008 9:59 pm


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