[Asterisk-Users] Re-Invite between SIP phones

[Al]

You are correct. T and t removed. Now reINVITE works.
Tks!

--- John Todd <jtodd at loligo.com> wrote:
> 
> I suspect you are using a Dial() statement that has
> something like 
> "T" or "t" on it, which will force the media path
> through Asterisk so 
> that Asterisk can listen for # keypresses.
> 
> Please include the full context of the dialing
> routine so it can be 
> examined.  Trim down a test to the absolute simplest
> form of a Dial 
> and try to see if reinvite works.
> 
> JT
> 
> 
> At 6:30 AM -0800 1/20/04, Al wrote:
> >
> >I'm trying to place calls between Cisco ATAs and
> >XLite clients. Calls go through perfectly.
> >
> >Both sides of the call negotiate the same CODEC
> >(G711a).
> >
> >I read that older Cisco ATA 186 firmwares don't
> >support reinvites but when capturing traffic there
> is
> >no Asterisk attempt to send the reinvite message.
> >
> >Al
> >
> >
> >--- "Low, Adam" <ALow at Prioritytelecom.com> wrote:
> >>  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
> >  > >
> 
> [People-  TRIM YOUR POSTS - there was like 6k worth
> of crap down here]
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