> -----Original Message----- > From: Vikram Rangnekar [mailto:[EMAIL PROTECTED] > Sent: 09 May 2005 11:26 > To: [email protected] > Subject: [Asterisk-Users] Connecting 20+ asterisk servers together > > > I have 20+ asterisk servers and need to network them together > so a phone on any of the servers can call a phone on any > other server without any trouble. > > I can think of IAX trunks between every server. So every > server will have an IAX trunk to every server and then prefix > bases routing in the dialplan for each server (I can give a > number to each server and use that as a prefix for that > server). But I think this is a maintainance nightmare and > also a very bad approch does anyone have any better ideas, > Also should the phones be able to send rtp between each other > or only through the Asterisk server since if its through the > asterisk server and say an IAX trunk then the max number of > calls can be controlled right. > > Can dundi or the switch statement help me get out of this mess ?
Am I right in saying this is for remote sites rather than 20 servers for load balancing reasons? As I am about to start hooking up a small number (< 5) sites together with Asterisk servers at each site and am not entirely sure of the best approach. I was thinking of having extensions for each site something like: 1xxx = site 1 2xxx = site 2 then for example server1 would have: switch => IAX2/user:[EMAIL PROTECTED]/context Matching on 2xxx. But this doesn't sound particularly elegant specially once you start trying to scale it. If you do get any other ideas I would be interested to know so that I can start this structure out properly. Again if this is remote sites, how are the phones going to talk directly to each other, VPN? Passing the RTP data over VPN direct to the phones will mean you don't get the benefits of the IAX trunking to reduce bandwidth which would be a shame. I would be interested to know how people find VPN's for passing audio, specially if IPSec etc is being used. I would imagine the quality is fairly bad on anything except the fattest of pipes. Is worth noting call monitoring will force the RTP to the servers regardless, I don't know if that's an issue for you? Sorry not a lot of answers there just questions :-D Cheers alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in a writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
