[asterisk-users] T1 problem (call using a .call file)

Fri Mar 20 09:53:55 CDT 2009

Don't really know the answer, but these are "givens":

1.	your phone is (most likely) in the same area code as the asterisk
installation
2.	NY is most likely not in the same area code.
3.	Even though the T1 is a dedicated digital service, the code that
handles all of this is/was written to process calls from analog sources for
backwards compatibility and therefore would have the timing issue handlers
in place even though they don't apply.

My research revealed that you might use an exception to stop this, but I
didn't really find a good example.  You could check viop-info.org or
whirlpool to see what they say.

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Pascal Bruno
Sent: Friday, March 20, 2009 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 problem (call using a .call file)

I still find it weird as even if it is a switch timing problem.  Because
when is it calling my phone all the time and that other area code it never
calls it.  Does that mean asterisk always complete my number in a certain
time frame, and the other number no?  Also I get the progress code 127
exactly after i move my call file to the outgoing folder, there is no delay,
I get it tthe same time I move the move.

And also why the call goes through when I put SIP/whatever in the callerid?
Does that mean asterisk get to complete the call in the time frame?

On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas <danny at debsinc.com> wrote:

You can also do a set variable in the call file.  I don't really know how to
do that, but you can probably find the command and syntax on voip-info.org
<http://voip-info.org/> .

The reason it works on certain numbers has to do with switch timing.  If *
can complete the call within a certain time frame, all is well.  If not, the
127 thing will bite you.

You would think we were past that type of thing, but I suppose not.

Another thing you might try is changing the 60 to 90 or so on your original
call file.

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Pascal Bruno
Sent: Thursday, March 19, 2009 4:42 PM 

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 problem (call using a .call file)

I dont want to change it within my extensions.conf, because I have many
dids, and change them on the fly according to the call i am making.  I have
a web interface where I fill a form that gets the number I am calling, the
caller id and context to go etc...

I dont want to keep editing extensions.conf and reload, I want to do it
directly in the call file.

What I dont understand is WHY it works on certain numbers and not all.  That
is a problem, it is not normal.

On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas <danny at debsinc.com> wrote:

GLOBAL_OUTBOUNDCID = XXXXXX in extensions.conf [globals] should do the trick

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com

[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Doug Lytle
Sent: Thursday, March 19, 2009 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 problem (call using a .call file)

Pascal Bruno wrote:
> Also very strange, when in my call file I change the callerid line to
> SIP/whatever like Danny said, the call go through, but I dont want
> that, because when I do so, it is displaying the main number on my T1
> account as caller id and I dont want that, I want to display one of my
> other DID as callerid.

Then change your caller-id within your dialplan, not the callfile.

Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."

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