asterisk users - May 2008 - Transfer

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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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