> 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 an idea. I'm playing with this now, but there's at least 1 case I'm having trouble recognizing: The call connects but then drops due to "unauthorized." It then only goes to the "h" extension and I don't get a chance to try again. Is there anyway to detect this? I have to cover all of the following cases: 1. VOIP IP address is not reachable. Goes to extension n+101 (seems to work as expected) 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? Yes I am using a correct username and password and getting this today (not from Voicepulse, from another provider). But there's also a moderate chance that during our systems' setup a name or password could be misspelled so I need to cover this case. 3. VOIP service connects but reports "all busy." Well this one is hard to test. But I can make the Zap channel busy. It goes to extension n+101 as expected, so I'll have to assume that a busy VOIP service does the same thing. I was trying to determine if the "t" or "h" extension would be useful for these but I think not. The timeout has to be set long enough for someone to actually answer (20-60 sec or whatever). The "h" is always visited at the end of the call, whether it was sucessful or not. Any other cases, or suggestions how to handle case #2?
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 about AGI yet but it may help solve this problem. --- Matt Lawson <matt@1control.com> wrote:> > 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 an idea. > > I'm playing with this now, but there's at least 1 case I'm having > trouble recognizing: > > The call connects but then drops due to "unauthorized." It then only > > goes to the "h" extension and I don't get a chance to try again. Is > there anyway to detect this? > > > I have to cover all of the following cases: > > > 1. VOIP IP address is not reachable. Goes to extension n+101 (seems > to > work as expected) > > 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? > > Yes I am using a correct username and password and getting this today > > (not from Voicepulse, from another provider). But there's also a > moderate chance that during our systems' setup a name or password > could > be misspelled so I need to cover this case.If your providers requires a pre-paid account the the account bvallance runs out then I gues you'd get "unauthorized". So this could be a real case that will happen> > 3. VOIP service connects but reports "all busy." Well this one is > hard > to test. But I can make the Zap channel busy. It goes to extension > n+101 as expected, so I'll have to assume that a busy VOIP service > does > the same thing.I get this from the stwo VOIP providers I use about 20% of the time. I guess they have only so much gateway hardware. Normally a quick re-dail does it.> > I was trying to determine if the "t" or "h" extension would be useful > > for these but I think not. The timeout has to be set long enough for > > someone to actually answer (20-60 sec or whatever). The "h" is > always > visited at the end of the call, whether it was sucessful or not. > > Any other cases, or suggestions how to handle case #2? > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users====Chris Albertson Home: 310-376-1029 chrisalbertson90278@yahoo.com Cell: 310-990-7550 Office: 310-336-5189 Christopher.J.Albertson@aero.org KG6OMK __________________________________ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus
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 trying to determine if the "t" or "h" extension would be useful > for these but I think not. The timeout has to be set long enough for > someone to actually answer (20-60 sec or whatever). The "h" > is always > visited at the end of the call, whether it was sucessful or not.Could Dial option 'g' (goes on in context if the destination channel hangs up) be of any help here? Florian