On Sat, Mar 7, 2009 at 2:20 PM, Johann Steinwendtner <steinwendt...@gmx.net>wrote:
> John Todd 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. > > > > You should be able to solve the callerid issue by using the sendrpid and > trustrpid prompts. > > Hans > > _______________________________________________ > -- 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 > Hans, Thanks for the tip - sendrpid and trustrpid is SOOOOO much more elegant!!!! Works like a charm!!! Thanks! Pedro
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