I am on Asterisk 1.4.23.1. What you propose is interesting. I will look into this ASAP to see if this will help. Thanks!<br><br><div class="gmail_quote">On Fri, Mar 6, 2009 at 4:23 PM, John Todd <span dir="ltr"><<a href="mailto:jtodd@digium.com">jtodd@digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
Just a suggestion: have you tried more recent versions of Asterisk<br>
with IAX2? I'm uncertain what version you're using, and if it's<br>
1.2.4, that's getting to be quite old and the problems that you<br>
reference may already be solved in more recent updates.<br>
<br>
In addition, if you're set on SIP, there are features in newer<br>
versions of Asterisk which allow you to both set and read SIP headers,<br>
so you can insert values in those headers between Asterisk instances<br>
which could then be used by the dialplan to split your calls apart<br>
into different contexts or behaviors.<br>
<br>
See function "SIP_HEADER" and application "SIPAddHeader" for the most<br>
recent versions of Asterisk.<br>
<br>
JT<br>
<div><div></div><div class="h5"><br>
<br>
On Mar 6, 2009, at 11:29 AM, tracinet wrote:<br>
<br>
> That stinks... We are migrating to SIP from IAX2 at the moment and<br>
> running into the same exact problem. No way to control the<br>
> destination context unless you use the "fromuser". Of course that<br>
> is rendering Caller ID useless as you pointed out.<br>
><br>
> I am still researching this though, if I find anything I will post<br>
> it here...<br>
><br>
><br>
> On Fri, Mar 6, 2009 at 2:13 PM, Adam Robins<br>
> <<a href="mailto:arobins@pharmacentra.com">arobins@pharmacentra.com</a>> wrote:<br>
> no<br>
><br>
><br>
> From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
> ] On Behalf Of tracinet<br>
> Sent: Friday, March 06, 2009 2:08 PM<br>
> To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
> Subject: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using<br>
> SIP<br>
><br>
><br>
><br>
> On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins<br>
> <<a href="mailto:arobins@pharmacentra.com">arobins@pharmacentra.com</a>> wrote:<br>
><br>
><br>
> I am switching from IAX2 to SIP for my inter-Asterisk transport due to<br>
> assorted quality issues following the 1.2.4 upgrade.<br>
><br>
> On the server that SENDS the call, I have the following in SIP.CONF:<br>
><br>
> [192.168.1.2_OB]<br>
> type=peer<br>
> fromuser=OB<br>
> host=192.168.1.2<br>
><br>
> And in EXTENSIONS.CONF<br>
><br>
> exten => 91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@192.168.1.2_OB)<br>
><br>
><br>
> On the RECEIVING Server in SIP.CONF:<br>
><br>
> [OB]<br>
> type=user<br>
> context=longdistance<br>
><br>
><br>
> I am not using a REGISTER statement on the receiving server.<br>
><br>
> My problem is that the only way I can seem to get the call delivered<br>
> into the proper SIP context on the receiving box is to use the<br>
> "fromuser=OB" on the sending machine. I tried using "username=OB",<br>
> but<br>
> then it delivers into the default context. I don't want to use<br>
> "fromuser" because it overrides the callerid.<br>
><br>
> Any suggestions?<br>
><br>
> Thanks,<br>
> Adam<br>
><br>
> The contents of this email message and any attachments are<br>
> confidential and are intended solely for addressee. The information<br>
> may also be legally privileged. This transmission is sent in trust,<br>
> for the sole purpose of delivery to the intended recipient. If you<br>
> have received this transmission in error, any use, reproduction or<br>
> dissemination of this transmission is strictly prohibited. If you<br>
> are not the intended recipient, please immediately notify the sender<br>
> by reply email and delete this message and its attachments, if any.<br>
><br>
><br>
> _______________________________________________<br>
> --Bandwidth and Colocation provided by Easynews.com --<br>
><br>
> Asterisk-Users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br>
><br>
><br>
> Did you ever get a resolution on this?<br>
><br>
><br>
> _______________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br>
> _______________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
</div></div>---<br>
<font color="#888888">John Todd <a href="mailto:email%3Ajtodd@digium.com">email:jtodd@digium.com</a><br>
Digium, Inc. | Asterisk Open Source Community Director<br>
445 Jan Davis Drive NW - Huntsville AL 35806 - USA<br>
direct: +1-256-428-6083 <a href="http://www.digium.com/" target="_blank">http://www.digium.com/</a><br>
</font><div><div></div><div class="h5"><br>
<br>
<br>
<br>
_______________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br>