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[asterisk-users] Transfer

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Adrian.Marsh at ubiqui...
Guest

Posted: Fri May 23, 2008 10:17 am Post subject: [asterisk-users] 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080523/9a56fb38/attachment.htm

sherwood.mcgowan at gm...
Guest

Posted: Fri May 23, 2008 11:25 am Post subject: [asterisk-users] Transfer

Adrian Marsh wrote:

Quote:

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

Adrian.Marsh at ubiqui...
Guest

Posted: Sun May 25, 2008 5:56 am Post subject: [asterisk-users] Transfer

Thanks Sherwood,

But how do I send back a 302, once I'm already in the dialplan (hasn't
asterisk already sent back a 200 OK by this point??)

Adrian

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sherwood
McGowan
Sent: 23 May 2008 17:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfer

Adrian Marsh wrote:

Quote:

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 ?

Quote:

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

Quote:

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

----------------------------------------------------------------------
--

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

http://www.google.com/search?hl=en&q=asterisk+302+redirect+sip&btnG=Sear
ch
I believe you're looking for a 302 Redirect? Sorry if you're not, but
that sounds like what you want
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Adrian.Marsh at ubiqui...
Guest

Posted: Sun May 25, 2008 6:26 am Post subject: [asterisk-users] Transfer

Quote:

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 ?

Quote:

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