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" <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]
 Sent: Tuesday, January 20, 2004 2:52 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Re-Invite between SIP
 phones


Already did that, but it's not working. Al

 --- "Low, Adam" <[EMAIL PROTECTED]> wrote:
 > canreinvite=yes within sip.conf entities ...
 >
 > -----Original Message-----
 > From: Al [mailto:[EMAIL PROTECTED]
 > Sent: Tuesday, January 20, 2004 2:06 PM
 > To: [EMAIL PROTECTED]
 > 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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