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

[Pascal Bruno]

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.
>
> 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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