[Asterisk-Users] Re-Invite between SIP phones

[Kannaiyan Natesan]

Can you please let me know how you made the test.
i will try to do at my end and check for the problems.

Thanks.

Kannaiyan

----- Original Message ----- 
From: "Al" <albor5000 at yahoo.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, January 20, 2004 4:00 PM
Subject: Re: [Asterisk-Users] Re-Invite between SIP phones


> 
> 
> What I would like to understand is in what situations
> reINVITEs are issued. 
> 
> Anyway, I got the following messages when trying to
> apply your patch.
> 
> patching file chan_sip.c
> Hunk #1 succeeded at 160 with fuzz 1.
> Hunk #2 succeeded at 365 with fuzz 1.
> Hunk #4 FAILED at 2253.
> Hunk #5 FAILED at 2291.
> Hunk #6 FAILED at 5019.
> Hunk #7 FAILED at 5168.
> Hunk #8 succeeded at 5911 with fuzz 1.
> Hunk #9 FAILED at 6245.
> Hunk #10 FAILED at 6397.
> patch unexpectedly ends in middle of line
> Hunk #11 FAILED at 6683.
> 7 out of 11 hunks FAILED -- saving rejects to file
> chan_sip.c.rej
> 
> Al
> 
> --- Kannaiyan Natesan <nkans at lycos.co.uk> wrote:
> > 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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