[Asterisk-Users] SIP Transfers (Possibly reinvite)

[Michael Graves]

On Tue, 10 Aug 2004 16:29:57 -0700, Chris Shaw wrote:

>----- Original Message -----
>From: "Christopher Jacob" <chris at jacob-solutions.com>
>To: <asterisk-users at lists.digium.com>
>Sent: Tuesday, August 10, 2004 3:57 PM
>Subject: [Asterisk-Users] SIP Transfers (Possibly reinvite)
>
>
>> Hey Folks,
>>
>> Is it possible to transfer an incoming call back out without a "trombone"
>> effect.
>>
>> For instance;
>>
>> Caller dials my broadvoice # --> Asterisk Answers and plays a menu --> the
>> caller selects an option --> asterisk transfers the call to my cell phone
>> via broadvoice and removes itself from the equation so I end up with...
>>
>> Caller --> Broadvoice --> Cell Phone
>>
>> Vs.
>>
>> Caller --> Broadvoice --> Asterisk --> Cell Phone
>>
>>
>> Any ideas on how this could work? I'm thinking it's something to do with
>> reinvite.
>>
>> Thanks
>>
>> Chris
>
>Unless you also have a PSTN connection (you didn't mention one) you will
>actually be doing something more like this:
>
>PSTN -> BroadVoice -> Asterisk -> BroadVoice -> Cellphone
>
>Not sure if re-invite really applies here... Basically what reinvite does is
>it uses the SDP information passed from a sip proxy to connect an IP phone
>that is connected to * directly to the calling party (in this case, the
>BroadWorks server), removing * from the equation... What you want to do is
>invite the broadvoice server back on itself which would create a loop and I
>don't think that will work...
>
>If you have enough bandwidth, the diagram shown above would work like a
>3-way call with * as the initiator and your cellphone as the 2nd leg... If
>you don't have the bandwidth, the other option would be to get an X100P card
>and have * dial your cellphone through POTS when someone dials that
>extension... This would remove you from a pure VoIP setup which is probably
>not what you want...

I used a related setup to have DISA access to my * box for making
overseas calls at cheaper rates. At present my setup works like this...

cell -> VoicePulse Connect DID -> Asterisk -> VoicePulse Connect -> UK
landline

Since * jumps out of the way once the call is connected I was hoping to
make the arrangement independent of my office bandwidth, etc. However
I've been having some reliability problems thus far, primarily around
random hangups. I was thinking of trying to use another provider for
the outbound leg so that VPC is not handling the call twice.

Michael
--
Michael Graves                           mgraves at pixelpower.com
Sr. Product Specialist                          www.pixelpower.com
Pixel Power Inc.                                 mgraves at mstvp.com

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