[asterisk-users] E1 configuration

[Flavio Miranda]

hi,
So, I think it depend of what environment are you setting up your link . In my case, E1 R2 Digital Brazil standard (Variant=br), I needed to change dahdi-channels parameter,chan_dahdi.conf , system.conf as well.

If you need I can send you such configuration.
good look!





Att,

 

Flavio Roberto Miranda

MSN:flaviormiranda at hotmail.com
Skype: flaviormiranda



Date: Tue, 26 Oct 2010 14:24:13 +0330
From: seighalani at gmail.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] E1 configuration

hi my friend

 would ou say what did you do for solving the problem? because i use a digium te121p and have many problems.


thanks in advance




On Mon, Oct 25, 2010 at 4:50 PM, Flavio Miranda <flaviormiranda at hotmail.com> wrote:






Sorry, thats right!!

I the nest email I will post here what I did in order to sove my problem!

Att,
 
Flavio Roberto Miranda
MSN:flaviormiranda at hotmail.com

Skype: flaviormiranda


 



Date: Sun, 24 Oct 2010 23:59:27 -0700
From: shakeel.abbas.qau at gmail.com
To: asterisk-users at lists.digium.com

Subject: Re: [asterisk-users] E1 configuration


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
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
Skype: flaviormiranda


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free is to know that  you have a different option



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