[Asterisk-Users] Routing SIP calls via URI

[Jeremy Wadhams]

Have you tried this guy's suggestion? (I have not, yet)

http://slacker.com/~nugget/projects/asterisk/page7

--JW
----- Original Message ----
From: Joao Pereira <joao.pereira at fccn.pt>
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Sent: Thursday, April 6, 2006 7:44:56 AM
Subject: Re: [Asterisk-Users] Routing SIP calls via URI

But is there a way of doing this without a prefix?

because people should dial without prefixes: "name at domain.pt" , not like:
"6name at domain.pt"

How can we make this without a prefix? something like:

if( !uri=~"@mydomain.pt" ){
 forward the all to the Internet
}

:)
Thanks
Joao Pereira


Shad Mortazavi wrote:

>Dear Group,
>
>I was able to fix this problem;
>
>The solution was to use a prefix to dial out. 
>
>The next challenge was to send the SIP Domain over IAX2!. I found that
>if I included @SIPDOMAIN it would break the IAX2 communications.
>
>exten => _6.,1,Dial(IAX2/bxxxxxx:yyyyyy at 192.X.y.x/${EXTEN}@SIPDOMAIN),
>breakes because @SIPDOMAIN is treated as the target context. You also
>can not include @Context after the @SIPDOMAIN.
>
>I created a new variable DS which was a concatenation of EXTEN and
>SIPDOMAIN separated by % and not @ and I was now able to pass this over
>IAX2;
>
>DS = EXTEN%SIPDOMAIN.
>
>exten => _6.,1,Dial(IAX2/bxxxxxx:yyyyyy at 192.X.y.x/${DS}).
>
>At the other end I used the CUT command and substring facilities in
>Asterisk to split DS by the % eliminator; I re-formed a new variable
>which was 
>
>DS = EXTEN at SIPDOMAIN.
>
>I can now pass calls from my internal Asterisk server to my external
>Asterisk server using IAX2 and then call any external VoIP number.
>
>Warm Regards
>
>Shad Mortazavi
>------------------------------
>Nexus Group Technical Manager
>n|m Nexus Management Inc
>
>-----Original Message-----
>From: Shad Mortazavi 
>Sent: Thursday, March 30, 2006 10:30 AM
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] Routing SIP calls via URI
>
>Dear Group;
>
>I can confirm that I have read through the three examples in
>www.voip-info.org. 
>
>These examples are excellent and address a couple of the questions. I
>have IAX2 working between several asterisk servers on our VPN and
>between the DMZ and our LAN. 
>
>Also
>
>exten => shad,1,Dial(IAX2/bxxxxxx:yyyyyy at 192.X.y.x/${EXTEN})
>
>This answers part of the question;
>
>However what I want to do is to send any outbound sip calls via our
>external SIP server.
>
>i.e;
>             VPN      LAN         IAX2    DMZ      Internet
>Internal UA <-------> Internal (*) <------> External (*)<------>
>ExternalUA
>
>We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
>for Voicemail, 2xxx for Meetme, etc. 
>
>Do I need to setup a prefix to dial the internet? And then route all
>calls to the External(*) based on this prefix?
>
>Thanks
>
>Shad Mortazavi
>------------------------------
>Nexus Group Technical Manager
>n|m Nexus Management Inc
>
>
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