Is this a bug

[john_usc]

I am using VERSION: 2.0.5-173 BUILD: 90320-0424 .
I set up the carriers and add a phone. Phone registers fine but when I try to dial a number through sip phone by dialing 91NXXNXXXXXX it says person is unavailable. The CLI gave me this



---------------------------------------------------------------------------------
-- Executing AGI("SIP/201-086406f8", "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/201-086406f8", "/18053647373||To") in new stack
Jul 16 20:43:08 WARNING[11473]: channel.c:2621 ast_request: No channel type registered for ''
Jul 16 20:43:08 NOTICE[11473]: 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/201-086406f8", "") in new stack
== Spawn extension (default, 918053647373, 3) exited non-zero on 'SIP/201-086406f8'
-- Executing DeadAGI("SIP/201-086406f8", "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

---------------------

here is my sip-vicidial.conf
---------------------

[general]
register => user:test@provider.com

[testvoip]
type=peer
host=provider.com
username=user
secret=test
qualify=yes
allow=all
canreinvite=no
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes
context=default


[201]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=201
secret=test
host=dynamic
dtmfmode=inband
qualify=1000
mailbox=201

-------

Info I entered in the carrier page was this right from the managers manual with ofcourse the obivous changes that I had to make.

This didnt work and the changes I added in the carriers page in the admin section were not writen to the extension.conf or extensions-vicidial.conf files neither were they written to sip or sip-vicidial.
Is there another way to do it or is this a bug or am I doing it wrong.
Thanks

[mflorell]

This is a dialplan issue, have you tried deleting the 91NXXNXXXXXX from the extensions.conf file and reloading asterisk?

[john_usc]

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

[gardo]

Try this getting started guide: http://vicidialnow.org/wiki/vnow/GettingStartedGuide . It might be able to help you.

[mflorell]

Is 91NXXNXXXXXX in your /etc/asterisk/extensions.conf file?

[williamconley]

-- 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 ''

"/17203334762||To" should have a "SIP/XXXXX/17203334762||To" where XXXXX is the sip context. Your system has not been told which context to use. (as further evidenced by No channel type registered for BLANK.)

The context is entered via the Dial(user:pass@jfk-primary.voicepulse.com/${EXTEN:2},,tTor), but is apparently being ignored.

Try finding (and deleting) the predefined _91NXXNXXXXXX in extensions.conf (it overrides and disables any loaded in via the GUI) or just use _81 instead of _91 and skirt the issue.

Also, the recommended standard format is not to directly define the context/user/pwd within the dialplan entry itself. Generally, this is done via the definition of a variable in the globals, and then the use of the variable in the dial plan instead of the actual user/pass@provider. In the GUI, you define a Globals String as "VARIABLE=SIP/context" and then you use "Dial(VARIABLE/...)". Then the system reaches into the context mentioned to get its user/pwd and anything else it needs.

But regardless, somehow this is being forgotten.

[THUFIR]

Also, the recommended standard format is not to directly define the context/user/pwd within the dialplan entry itself. Generally, this is done via the definition of a variable in the globals, and then the use of the variable in the dial plan instead of the actual user/pass@provider. In the GUI, you define a Globals String as "VARIABLE=SIP/context" and then you use "Dial(VARIABLE/...)". Then the system reaches into the context mentioned to get its user/pwd and anything else it needs.

But is the global string really global or is it just a variable for that context?

[williamconley]

But is the global string really global or is it just a variable for that context?

Global string exists outside of contexts. Look in the sip-vicidial.conf file and you will see that it is declared outside of any context. It can also be used in any context. Thus the phrase "global".

[THUFIR]

thanks. It was my understanding that global variables should be declared in the [globals] context. Ok.

[williamconley]

thanks. It was my understanding that global variables should be declared in the [globals] context. Ok.

you should not be editing the file manually. never do that. the global variables are generated by the system from the global variables field in the carrier settings.