Hello Joshua, On 04/25/2016 at 12:35 PM, Joshua Colp wrote: > Michael Maier wrote: >> Hello! >> >> I encounter the following problem (asterisk 11 and 13) with Teconisy as >> trunk provider with enabled qualify and default t1min (100ms): >> >> Teconisy most often doesn't answer the first invite before asterisk >> default t1min ended. Therefore asterisk sends one more invite. This >> second invite is answered by Teconisy with >> >> status 486 - Request terminated - Channel limit exceeded. >> >> (The second invite obviously is interpreted as second, new call). >> >> Asterisk therefore cancels the first(!!) call - but Teconisy proceeds >> with exactly this first call (which now can't be handled by asterisk any >> more). >> >> For me, there are two problems in asterisk at this point: >> >> 1. The VoIP standard defines 500ms for t1 - using this standard value >> for t1min works fine with Teconisy, too. t1min should be always >> 500ms - it prevents a completely blocked line! > > The standard actually allows you to ignore this and use the estimated > round trip time, which chan_sip will derive from the time it takes an > OPTIONS packet to go out and get a response. The t1min just enforces a > minimum. > >> 2. Why does asterisk stop the call completely after the second invite, >> which is canceled by Teconisy? It should be ignored because it >> means, that the first invite is already processed by Teconisy. > > The 486 response code is actually for indicating busy. The 4XX series is > also for client failure, which tear down the session. It can't be > ignored. This would break things. > > The retransmission of an INVITE shouldn't result in multiple sessions > being set up, the other side should treat it as a retransmission. This > is a bug on their side. Your adjustment of the T1 just makes it so you > don't allow it to happen. > > I'd say the problem isn't with Asterisk but with the remote side. Since > you can configure things to work that's great but I don't see any code > changes we can do.
thanks for clarification! Besides that - I'm really wondering why others don't face this problem - I hardly can believe that I'm the only one affected. Regards, Michael -- _____________________________________________________________________ -- 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
