[Asterisk-Users] Re-Invite between SIP phones

[Kannaiyan Natesan]

Hi,

   I think canreinvite=yes won't work in most of the situations.
   I have implemented Redirect SIP 300 Message to redirect to the SIP
address you speficy in the sip.conf.

   Where you can have ,


    register => username:password at domain.com/extension

   [extension]
    redirect=yes
    redirecturi=sip:12345 at domain1.com
    redirecturi=sip:56442 at domain2.com
    ...

    will make to redirect to all the URI's yu specify in the sip.conf.  I'm
also working on this so that it can get the redirections from the database
rather than reloading asterisk all the time when you modify the redirection
uri.

   You can check through that.

   http://bugs.digium.com/bug_view_page.php?bug_id=0000879

   Message transmission is alright, but for some reason it is not working.
Can you test with yours and let me know where is the problem, I will modify
the code once you get the clue where is the problem on it. If successfully
please send me the sip debug message and I will just make sure it works for
all.

Kannaiyan


----- Original Message -----
From: "Low, Adam" <ALow at Prioritytelecom.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, January 20, 2004 2:00 PM
Subject: RE: [Asterisk-Users] Re-Invite between SIP phones


> I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats
happening because it works great for me and always has but I guess it also
requires support on the end-points and possibly (assuming non-cisco enviro)
there maybe an option that needs to be configured on your phones/gateways.
>
> Please provide more information on your setup ...
>
> -----Original Message-----
> From: Al [mailto:albor5000 at yahoo.com]
> Sent: Tuesday, January 20, 2004 2:52 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Re-Invite between SIP phones
>
>
> Already did that, but it's not working.
> Al
>
> --- "Low, Adam" <ALow at Prioritytelecom.com> wrote:
> > canreinvite=yes within sip.conf entities ...
> >
> > -----Original Message-----
> > From: Al [mailto:albor5000 at yahoo.com]
> > Sent: Tuesday, January 20, 2004 2:06 PM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Re-Invite between SIP
> > phones
> >
> >
> > Anybody knows what do I need to tell Asterisk
> > to issue a re-INVITE between two SIP phone to avoid
> > having the media going through the server?
> >
> > Tks,
> > Al
> >
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otherwise protected from disclosure and may include proprietary information.
If you are not the intended recipient, please telephone or email the sender
and delete this message and any attachment from your system. If you are not
the intended recipient you must not copy this message or attachment or
disclose the contents to any other person
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