[Asterisk-Users] Re: failover (was Re: voicepulse)

2004-01-14 Thread Matt Lawson
But this is not to say _you_ can't built a reliable VOIP based system. Get _two_ providers and set up your dial plan in extensions.conf to fail over if one service fails to connect to dial via the next one and finally if both fail use pstn. your users will see a system the just works. Now there's

[Asterisk-Users] Re: failover (was Re: voicepulse)

2004-01-14 Thread Matt Lawson
OK, so I answered my own question. Turns out case #2 just goes to extension 2. Still trying to figure out the optimum arrangement so I don't have an inordinate number of extensions. Maybe like this: 1. First outgoing try 2. Second outgoing try 3. Third ougoing try 4. Play a message

Re: [Asterisk-Users] Re: failover (was Re: voicepulse)

2004-01-14 Thread Chris Albertson
I'm having the same concerns. What we REALLY need is the ability to test the exact nature of the problem. OK We could use SER to front end SIP calls but Asterisk should report the problem and allow the dial plan to test it. It's a needed missing feature in * What about AGI? I don't know much

RE: [Asterisk-Users] Re: failover (was Re: voicepulse)

2004-01-14 Thread Florian Overkamp
Hi, -Original Message- 2. VOIP service answers but refuses with call with unauthorized. It just goes to the h extension Is there any watch to catch this failure? Perhaps put a timer on it and say if the call was less than 5 seconds or something try the next one? I was