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