Taking a look at it, I'm not sure if that will do what I need; I need it to
initially connect with the PBX and then be re-invited to continue the same
call directly.
My SIP trunk, Unlimitel, recognizes two phones within the same network and
connects the phones directly, so I need a way to simulate that. Is there a
way to manipulate the header information in the xml scripts?
On Mon, Apr 27, 2009 at 11:18 AM, mayamatakeshi <[email protected]>wrote:
> On Mon, Apr 27, 2009 at 11:37 PM, Paul Fugere <[email protected]>
> wrote:
> > I am trying to get SIPP to simulate a re-invite request and I am unsure
> on
> > how to exactly do this.
> >
> > What I want to do is have the call contact a number on an Asterisk PBX,
> and
> > once the call is established, I would like to have it receive/send a
> > re-invite and talk directly to the phone instead of having to pass
> through
> > the Asterisk server. Is this possible with SIPP or is this a limitation
> of
> > the software?
>
> Hello,
> I never had to use SIPp to talk to entities other than the initial server.
> Some threads in the forum hint that SIPp doesn't follow Via and Contact
> headers.
> If that's the case, maybe you can use the setdest action to change the
> destination of the call:
> http://sipp.sourceforge.net/doc/reference.html#setdest
>
> regards,
> takeshi
>
--
Paul Fugere
[email protected]
2T Computer Engineering
Engineering Society 'A'
Iron Warrior Advisory Board - A-Soc-At-Large
Bowling Director
Novelties Director
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POETS Programmer
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