Thanks for your help<br><br>
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<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Don’t really know the answer, but these are “givens”:</span></font></p>
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<li style="COLOR: navy"><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">your phone is (most likely) in the same area code as the asterisk installation</span></font></li></ol></div>
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<div>My phone has a different area code than the asterisk installation. The asterisk box is in FL and I can call a number in MN but not the 201 or many others</div>
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<li style="COLOR: navy"><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">NY is most likely not in the same area code.</span></font></li></ol></div></div></blockquote>
<div>I agree but I could call a MN cell phone for example which works all the time</div>
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<li style="COLOR: navy"><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">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.</span></font></li>
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<p style="MARGIN-LEFT: 0.25in"><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">My research revealed that you might use an exception to stop this, but I didn’t really find a good example. You could check <a href="http://viop-info.org/" target="_blank">viop-info.org</a> or whirlpool to see what they say.</span></font></p>
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<p><b><font face="Tahoma" size="2"><span style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">From:</span></font></b><font face="Tahoma" size="2"><span style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b><span style="FONT-WEIGHT: bold">On Behalf Of </span></b>Pascal Bruno<br>
<b><span style="FONT-WEIGHT: bold">Sent:</span></b> Friday, March 20, 2009 9:39 AM
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<div class="h5"><br><b><span style="FONT-WEIGHT: bold">To:</span></b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b><span style="FONT-WEIGHT: bold">Subject:</span></b> Re: [asterisk-users] T1 problem (call using a .call file)</div>
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<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">I still find it weird as even if it is a switch timing problem. Because when is it calling my phone <strong><b><font face="Times New Roman">all the time </font></b></strong>and that other area code it <strong><b><font face="Times New Roman">never </font></b></strong>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.</span></font></p>
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<p style="MARGIN-BOTTOM: 12pt"><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">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?</span></font></p>
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<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">On Thu, Mar 19, 2009 at 5:54 PM, Danny Nicholas <<a href="mailto:danny@debsinc.com" target="_blank">danny@debsinc.com</a>> wrote:</span></font></p>
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<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">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 <a href="http://voip-info.org/" target="_blank">voip-info.org</a>.</span></font></p>
<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">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.</span></font></p>
<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">You would think we were past that type of thing, but I suppose not.</span></font></p>
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<p><font face="Arial" color="navy" size="2"><span style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Another thing you might try is changing the 60 to 90 or so on your original call file.</span></font></p>
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<p><b><font face="Tahoma" size="2"><span style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">From:</span></font></b><font face="Tahoma" size="2"><span style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b><span style="FONT-WEIGHT: bold">On Behalf Of </span></b>Pascal Bruno<br>
<b><span style="FONT-WEIGHT: bold">Sent:</span></b> Thursday, March 19, 2009 4:42 PM </span></font></p>
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<p><font face="Tahoma" size="2"><span style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"><br><b><span style="FONT-WEIGHT: bold">To:</span></b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b><span style="FONT-WEIGHT: bold">Subject:</span></b> Re: [asterisk-users] T1 problem (call using a .call file)</span></font></p>
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<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">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...</span></font></p>
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<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">I dont want to keep editing extensions.conf and reload, I want to do it directly in the call file.</span></font></p></div>
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<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">What I dont understand is WHY it works on certain numbers and not all. That is a problem, it is not normal.</span></font></p>
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<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">On Thu, Mar 19, 2009 at 5:24 PM, Danny Nicholas <<a href="mailto:danny@debsinc.com" target="_blank">danny@debsinc.com</a>> wrote:</span></font></p>
<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">GLOBAL_OUTBOUNDCID = XXXXXX in extensions.conf [globals] should do the trick</span></font></p>
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<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt"><br>-----Original Message-----<br>From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a></span></font></p>
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<p style="MARGIN-BOTTOM: 12pt"><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Doug Lytle<br>
Sent: Thursday, March 19, 2009 3:45 PM<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>Subject: Re: [asterisk-users] T1 problem (call using a .call file)</span></font></p></div>
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<p><font face="Times New Roman" size="3"><span style="FONT-SIZE: 12pt">Pascal Bruno wrote:<br>> Also very strange, when in my call file I change the callerid line to<br>> SIP/whatever like Danny said, the call go through, but I dont want<br>
> that, because when I do so, it is displaying the main number on my T1<br>> account as caller id and I dont want that, I want to display one of my<br>> other DID as callerid.<br><br><br>Then change your caller-id within your dialplan, not the callfile.<br>
<br>Doug<br><br><br>--<br><br>Ben Franklin quote:<br><br>"Those who would give up Essential Liberty to purchase a little Temporary<br>Safety, deserve neither Liberty nor Safety."<br><br><br>_______________________________________________<br>
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