For the benefit of the archives, my problem was a simple one. I hadn't forwarded the IAX port on the router of the remote * server connection, so when voip provider was trying to connect directly to the remote * server, it couldn't.
Hurray for wasting an entire day over a simple silly little thing ;-) Mat > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Mat Stace > Sent: 06 November 2006 17:42 > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Asterisk servers being greedy and > not letting goof the media path. (using IAX2 channels) > > > Evening everyone (obviously depends on when you're readin > this, but hey). > > I'm trying to set up a multi * server situation, and am > falling over at the second server, and after a day of google > etc, have come up against somewhat of a brick wall. > > I can make calls each way between the two servers no problem, > and can include the required extension at the remote * server > as part of my main incoming dialplan. My problem comes with * > attempting to pass the media path to the other server. > > What is happening is: > > Incoming call from iax2 provider to main * server > --> dial sip extension on main * server > --> setup IAX2 channel to remote * server (which then rings > extension) > > Pickup call on extension on remote * server > --> main server sip extension stops ringing > --> ast console on main server I get : > > ------------------------------------------------- > -- Attempting native bridge of IAX2/voipprovider/6 and > IAX2/remote*server/7 > -- Channel 'IAX2/voipprovider/6' unable to transfer > -- Channel 'IAX2/remote*server/7' unable to transfer > ------------------------------------------------- > > In the user/friend declarations (user for incoming voip > provider, friend for remote * server) in the two iax.conf > files I have notransfer=no, and also up in the [general] > section of the iax.conf. > > The problem is that when remote * user answers the phone, and > then transfers the call to an extension on the main * server, > there is massive (ie 2 > seconds) delay, and using IAX2 show channels at the two > consoles, the call is doing the following: > > PSTN -> VOIP PROVIDER -> main * server -> remote * server -> > main * server > -> SIP extension on main * server. > > Anyone have any ideas on how to make the * servers give up > the media path? > > Cheers > > Mat > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.430 / Virus Database: 268.13.28/518 - Release > Date: 04/11/2006 17:30 > > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users