[asterisk-users] Get Asterisk to redirect a SIP INVITE

[Salvatore Giudice]

I don't think you can do that. You can easily issue a 302 with something
like SER or OpenSER. I believe the only thing Asterisk can do is receive a
call on the initial URI and open a channel to the destination and connect
them. Media could pass directly between those two points but your Asterisk
box would still have to participate in the signaling. Think of Asterisk as a
B2BUA instead of a SIP call router/response system.

--------------------------------------------------
Salvatore Giudice
Salvatore.Giudice at VoIPSecurityTraining.com

VoIP Security Training, LLC
http://VoIPSecurityTraining.com

848 N. Rainbow Blvd. #1676
Las Vegas, NV 89107
Phone: (617) 959-7625
Fax: (214) 279-2906


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Yuan LIU
Sent: Thursday, May 03, 2007 6:18 PM
To: asterisk-users at lists.digium.com
Subject: RE: [asterisk-users] Get Asterisk to redirect a SIP INVITE

>From: "CSB" <cameron.beattie at appsteam.co.nz>
>Date: Thu, 3 May 2007 21:51:02 +1200
>
>I want to get Asterisk to redirect an incoming SIP INVITE to another SIP 
>URI. I was looking at the Transfer application but it seems to

You may want to elaborate the requirement.  How is the incoming INVITE 
initiated?  Is the originator a user in your system?  Does the other URI 
represent a peer? etc.

Yuan Liu

>be broken (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483). 
>Is there an alternative way to do this on Asterisk 1.2.18?
>
>Regards
>
>Cameron


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