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 <[email protected]> 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:* [email protected] [mailto: > [email protected]] *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 <[email protected]> wrote: > > GLOBAL_OUTBOUNDCID = XXXXXX in extensions.conf [globals] should do the > trick > > > -----Original Message----- > From: [email protected] > > [mailto:[email protected]] 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." > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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