Thinking about it, since the Trunk is initiating the re-invite, I might be
going about the hard way by trying to initiate it from my SIPP.
Is it possible to spoof it so that SIPP can act as a registered phone on my
PBX and contact an external phone number? I remember trying this and I was
receiving errors because it was discarding a Register packet.
On Mon, Apr 27, 2009 at 11:49 AM, mayamatakeshi <[email protected]>wrote:
> On Tue, Apr 28, 2009 at 12:32 AM, Paul Fugere <[email protected]>
> wrote:
> > 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.
>
> I would expect it to work if you force SIPp to do the first re-invite
> (after using setdest).
>
> > Is there a way to manipulate the header information in the xml scripts?
>
> Yes. You can use action ereg to get data from any header or from the
> whole message and inject the data in subsequent messages or actions
> (that means, you could extract the ipaddress/port from the header
> Contact and use them when calling action setdest).
>
> > 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
> > POETS Manager
> > POETS Programmer
> >
>
--
Paul Fugere
[email protected]
2T Computer Engineering
Engineering Society 'A'
Iron Warrior Advisory Board - A-Soc-At-Large
Bowling Director
Novelties Director
POETS Manager
POETS Programmer
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