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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>In that case, you’re going to have to talk to your provider.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>They SHOULD be able to easily send the DID with the call…<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>-Dave<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>John
Taylor<br>
<b>Sent:</b> Tuesday, December 15, 2009 5:17 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] multiple sip trunks<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>I thought so- the fact the
server has 20 different registry entries to 20 different account all at the
same ITSP shouldn't matter?<br>
<br>
Can't see any DDI info in the SIP headers unfortunately :(<br>
<br>
John<o:p></o:p></p>
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<p class=MsoNormal>2009/12/14 meetmecall <<a href="mailto:info@meetmecall.nl">info@meetmecall.nl</a>><o:p></o:p></p>
<p class=MsoNormal>The easiest solution to deal with this is to have one
context with<br>
different extensions for the different numbers and route the incoming<br>
calls from there. It should look something like this (not a tested<br>
piece of asterisk script, just an example to give the idea).<br>
<br>
Hope it helps :-)<br>
<br>
<br>
Erik de Wild<br>
<br>
[all_trunks]<br>
<br>
exten => 31592123456,1,Goto(trunk1,s,1)<br>
exten => 31592123457,1,Goto(trunk1,s,1)<br>
exten => 31592123458,1,Goto(trunk1,s,1)<br>
<br>
exten => 31592123333,1,Goto(trunk2,s,1)<br>
exten => 31592123334,1,Goto(trunk2,s,1)<br>
exten => 31592123335,1,Goto(trunk2,s,1)<o:p></o:p></p>
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<p class=MsoNormal><br>
<br>
<br>
On 14 dec 2009, at 10:39, Olle E. Johansson wrote:<br>
<br>
><br>
> 11 dec 2009 kl. 23.21 skrev John Taylor:<br>
><br>
>> I have multiple trunks to the same ITSP. Incoming calls to any trunk<br>
>> go to the last "incoming" label defined in those trunks'
contexts in<br>
>> sip.conf.<br>
>><br>
>> My ITSP insists on insecure=very in the trunk context; is this the<br>
>> cause?<br>
>><br>
> This is an effect of the Asterisk architecture. We've had many<br>
> discussions on how to change it, but right now the peer matching on<br>
> IP/Port can't separate various instances from each other, since they<br>
> all have the same IP/port. Asterisk simply goes for the first match,<br>
> which happens to be the last entry with the IP/port in the sip.conf<br>
> file.<br>
><br>
> /Olle<br>
> _______________________________________________<br>
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<p class=MsoNormal><o:p> </o:p></p>
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