Done that. I just found that it's a documented bug. http://bugs.digium.com/view.php?id=7458 . Which is a bad thing, since it doesn't mean it's anything I can fix at the moment.
Gary On 8/28/06, Stephan Monette <[EMAIL PROTECTED]> wrote:
Try with the canreinvite=no in the sip.conf file. Maybe the SIP devices of the agent are not SIP reinvite compatible. Stephan Monette Unlimitel inc. Tel.: 1 (877) 464-6638, x221 Fax.: (613) 443-7773 -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen Sent: Monday, August 28, 2006 4:59 PM To: [email protected] Subject: [on-asterisk] Asterisk gets hosed on call transfer out of queue. I have agents in a queue, and when they receive a call, answer it, and then transfer it to someone else (either internal or external), the call is dropped and asterisk gets hosed, to the point where the queues dont work anymore, I can't get a listing of logged on agents or sip channels. The last thing I see when the call is transferred is: SIP/xxxxxxxxxx-9a13 is making progress passing it to Local/[EMAIL PROTECTED],2 Can anyone give my ideas at what I should look at? Thanks, Gary --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
