[asterisk-users] E1 configuration

[ABBAS SHAKEEL]

although I don't need the solution personally But would like to request you
that instead of posting "forget it" ..... if you post the solution to the
problem it will be more helpful.
In case some one else faces the same problem he can use your solution....

Good luck

On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda
<flaviormiranda at hotmail.com>wrote:

>  Forget it !!
>
>
>  After several  attempts, I have solved !!!
>
>
> Att,
>
> Flavio Roberto Miranda
> MSN:flaviormiranda at hotmail.com <MSN%3Aflaviormiranda at hotmail.com>
> Skype: flaviormiranda
>
>
>
> ------------------------------
> From: flaviormiranda at hotmail.com
> To: asterisk-users at lists.digium.com
> Date: Sun, 24 Oct 2010 22:28:16 -0200
> Subject: [asterisk-users] E1 configuration
>
>
> Hi all,
>
>   Please, anybody  that have some knowllege   about E1 configuration could
> give some guidance about it?
>
> I trying to set an Asterisk with E1 CAS signalling and  everything looks
> good, but when I try to go out with calls I receive the follow message:
>
> == Using SIP RTP CoS mark 5
>     -- Executing [21341400 at local:1] Dial("SIP/4804-00000000",
> "DAHDI/g11/21341400,,t") in new stack
>   == Everyone is busy/congested at this time (1:0/0/1)
>   == Spawn extension (local, 21341400, 2) exited non-zero on
> 'SIP/4804-00000000'
>
> The boad  has succesfully installed:
>
> Digium Wildcard TE110P T1/E1 Card 0      OK      0      0      0      CAS
> HDB3          0 db (CSU)/0-133 feet (DSX-1)
>
> the channels are correct and mfcr2 too, but the calls dont go out.
>
> Thanks for any help.
>
>
>
> Att,
>
> Flavio Roberto Miranda
> MSN:flaviormiranda at hotmail.com <MSN%3Aflaviormiranda at hotmail.com>
> Skype: flaviormiranda
>
>
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-- 
Best Regards
Shakeel Abbas
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