[Asterisk-Users] repost: SIP/Asterisk behavior

[Lal, Deepak (Contractor)]

In my setup, When asterisk receives a SIP INVITE request, the request URI in my
case is 5557777 at asterisk.ip.address <mailto:5557777 at asterisk.ip.address>  . The
SIP INVITE PDU's  message header also contains a To: field. In my case the To:
field is 4121891 at asterisk.ip.address <mailto:4121891 at asterisk.ip.address>   . It
seems that asterisk "accepts" the request-URI number as the called number
(5557777) and invokes the rule for the extension 5557777 as defined in my
extension.conf file. 

 

But, as per SIP, the call is really intended for 4121891 and not 5557777.  I'd
like asterisk to "go to"  the rule for extension 4121891 when it receives the
SIP request. Is this possible? I imagine that Asterisk would have to act as a
proxy to do this. 

 

Thanks in advance - DL

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040330/4880de8b/attachment.htm