[asterisk-users] What would cause a drop between two asterisk systems?
We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels. Lately we've been getting a disconnected calls. Keeping the consoles running it doesn't seem to be the PRI initiating the hangups, as I'll when I see hangups intiiated on the backend / PRI side: -- Span 2: Channel 0/21 got hangup request, cause 16 Instead, I'm seeing == Spawn extension (outbound, (dialed #), 3) exited non-zero on 'IAX2/asterisk-frontend2-603' -- Hungup 'IAX2/asterisk-frontend2-603' Which indicates the frontend initiated a hangup. But on the frontend I'm seeing auto fallthroughs to the h extension, which only happens if the hangup is initiated from the backend: -- Auto fallthrough, channel 'SIP/phone1-0167' status is 'ANSWER' (h extension stuff follows) If that side was initiating the hangup, I'd just see a jump to the h extension, with no auto fallthrough. So it looks like there may be a communication interruption between the front and backends. The problem is this happens intermittently, so I can't reproduce it reliably. I've held open a call for 30+ minutes and not run into the problem, while someone's been on a call for 7 minutes and this happens. It doesn't seem feasible to constantly run IAX2 debugs from the console on any open call - does anyone have suggestions on how to troubleshoot this? Weirdly enough, this only seems to happen when users dial into conference bridges (not local) such as WebEx and GoToMeeting, but that might just be because of the length of those calls. Will tweaking things like the IAX2 jitter buffer help? The two systems are barely four hops apart with an average of .2 ms ping times between them on a very resilient network (two of those hops are through core transports). I've never seen ping loss between them, even when running ping tests for hours during heavy call volume periods. The loads on the machines are minimal - never seen the load go above .10 during normal operation. But it does seem like something between them is making them drop calls. hose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What would cause a drop between two asterisk systems?
On Tue, Mar 5, 2013 at 2:32 PM, Hose hose+aster...@bluemaggottowel.comwrote: We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels. IAX is buggy. We've never seen a reliable system using it. We've given up on it. I'd try SIP. Easy to do, no real reason not to. Check all of the networking involved. Leave a ping test running between the systems constantly, then see if it dropped packets when you get a dropped call. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What would cause a drop between two asterisk systems?
Carlos Alvarez wrote: On Tue, Mar 5, 2013 at 2:32 PM, Hose hose+aster...@bluemaggottowel.com mailto:hose+aster...@bluemaggottowel.com wrote: We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels. IAX is buggy. We've never seen a reliable system using it. We've given up on it. I have seen this assertion from time to time, but never any real details There is a world wide network of users who communicate using IAX, and many with PSTN service from providers using IAX. with no complaints Can someone please provide meaningful details on what buggy really means? Rather than such a sweeping condemnation. If it is so buggy, why isn't it either fixed or discontinued? It certainly is much less prone to hacking and abuse than SIP. Probably not due to the protocol design as much as it isn't as universal John Novack -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What would cause a drop between two asterisk systems?
What you say...John Novack (jnov...@stromberg-carlson.org): Carlos Alvarez wrote: On Tue, Mar 5, 2013 at 2:32 PM, Hose hose+aster...@bluemaggottowel.com mailto:hose+aster...@bluemaggottowel.com wrote: We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels. IAX is buggy. We've never seen a reliable system using it. We've given up on it. I have seen this assertion from time to time, but never any real details There is a world wide network of users who communicate using IAX, and many with PSTN service from providers using IAX. with no complaints Can someone please provide meaningful details on what buggy really means? Rather than such a sweeping condemnation. If it is so buggy, why isn't it either fixed or discontinued? It certainly is much less prone to hacking and abuse than SIP. Probably not due to the protocol design as much as it isn't as universal John Novack I'll keep the SIP option open - never really considered it actually, just figured the go-to protocol for connecting two * boxes together would be IAX2. It'd be good for troubleshooting purposes at least to narrow down issues and compare results. If anyone else has suggestions, I'm all ears. hose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What would cause a drop between two asterisk systems?
On 6/03/2013, at 9:06 AM, John Novack jnov...@stromberg-carlson.org wrote: Carlos Alvarez wrote: On Tue, Mar 5, 2013 at 2:32 PM, Hose hose+aster...@bluemaggottowel.com wrote: We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels. IAX is buggy. We've never seen a reliable system using it. We've given up on it. IAX seems easy to me We run interoffice from NZ to Australia and many systems in between. No issues at all Cheers Duncan I have seen this assertion from time to time, but never any real details There is a world wide network of users who communicate using IAX, and many with PSTN service from providers using IAX. with no complaints Can someone please provide meaningful details on what buggy really means? Rather than such a sweeping condemnation. If it is so buggy, why isn't it either fixed or discontinued? It certainly is much less prone to hacking and abuse than SIP. Probably not due to the protocol design as much as it isn't as universal John Novack -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
