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

John Todd jtodd at digium.com
Fri Mar 6 15:23:11 CST 2009


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
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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