[asterisk-users] Transfer

[Sherwood McGowan]

Adrian Marsh wrote:
>
> Hi All,
>
> In my old telco days (SS7), if I was wanting to hand back a call to 
> the network for transfer to a different PSTN number, there was a 
> specific SS7 action I could take, which send the call back to the 
> network, which in turn then routed the call appropriately. It added a 
> transfer-number into the SS7 headers so that the originating number, 
> dialed number and transfer number all stayed to specs, and everyone 
> was happy.
>
> In SIP/Asterisk, it seems that I have to hairpin/trombone the call (to 
> have at least the control packets go via my SIP server), and use a 
> Dial out to the far end.
>
> So – is there a way of handing the call back to the network in asterisk ?
>
> My detailed problem is this: When a call comes in, I want to send it 
> onto users mobiles, if I hairpin the call that’s OK, except the CLI 
> needs to be that of the originator (from the USERS point of view) so 
> they can decide if they want to accept the call.
>
> Here in the UK, this is where the issues begin… the carriers here 
> don’t like it if your sending CLI for other countries, that don’t 
> match what they think they should receive from that connecting 
> carrier. Eg, if a call coming to them is 13 digits, but they only 
> expect 11 from that carrier, then they cut the digits. This turns a US 
> originated call into a Southampton UK originated call!
>
> So I was hoping that handing the call back to the network in the 
> traditional sense would make it their problem and not mine… lol
>
> Thanks,
>
>
> Adrian
>
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http://www.google.com/search?hl=en&q=asterisk+302+redirect+sip&btnG=Search
I believe you're looking for a 302 Redirect? Sorry if you're not, but 
that sounds like what you want