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

[tracinet]

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 <jtodd at 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
> > <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.
> >
> > Any suggestions?
> >
> > Thanks,
> > Adam
> >
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> >
> >
> > Did you ever get a resolution on this?
> >
> >
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> ---
> John Todd                       email:jtodd at digium.com<email%3Ajtodd at 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/
>
>
>
>
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