[Asterisk-Users] Re-Invite between SIP phones

[Al]

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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> otherwise protected from disclosure and may include
> proprietary information.
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> 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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