[asterisk-users] Que on A2Billing

Guillermo Salas M. gsalas at manta.telconet.net
Fri Jun 15 16:18:14 CDT 2007


On Fri, 2007-06-15 at 17:01 -0400, Nitesh Divecha wrote:
> Thanks man... That really helped me to move couple of steps. Now I see
> the incoming calls are going in proper direction... I know I am still
> missing a small piece here... I did ADD the Destination as a
> SIP/2486543210, assigned the card number, enabled VOIP_CALL, and
> enabled Active. 
> 


2486543210 is your card number?


> When I dial the DID number, on the *CLI it shows the following: -
> 
> a2billing.php|1|did: bug
>     -- AGI Script Executing Application: (DIAL) Options:
> (SIP/2486543210|60|HL(3600000:61000:30000))
>     -- Limit Data for this call: 
>     -- - timelimit     = 3600000
>     -- - play_warning  = 61000
>     -- - play_to_caller= yes
>     -- - play_to_callee= no
>     -- - warning_freq  = 30000
>     -- - start_sound   = UNDEF
>     -- - warning_sound = timeleft 
>     -- - end_sound     = UNDEF
> Destroying call '0e6e480117744f926cea9bcd6bb47104 at 69.134.49.17'
> Jun 15 16:41:34 NOTICE[15346]: app_dial.c:1069 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 3 - No route to destination) 
>   == Everyone is busy/congested at this time (1:0/0/1)
> 

I think that 2486543210 is not a customer, card number or SIP/IAX2
friend, maybe is PSTN number. To redirect the call to any PSTN number
you must need to set "voip call" to inactive and set the destination
number to 2486543210.


> I bet I am missing something in extension.conf correct? I dont see any
> examples in my package.
> 


The context is fine don't worry about it.


> Any suggestion... Thanks once again... 


Regards,

-- 
Guillermo Salas M.
Telconet S.A.
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