[asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP

[tracinet]

On Sat, Mar 7, 2009 at 2:20 PM, Johann Steinwendtner
<steinwendtner at 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
> >> <arobins at pharmacentra.com> wrote:
> >> no
> >>
> >>
> >> From: asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at 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
> >> <arobins at 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/${EXTEN}@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
>
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Hans,
Thanks for the tip - sendrpid and trustrpid is SOOOOO much more elegant!!!!

Works like a charm!!!

Thanks!

Pedro
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