[asterisk-users] Second invite after 100ms (with default t1min=100) --> canceled call problem!

Joshua Colp jcolp at digium.com
Mon Apr 25 05:35:38 CDT 2016


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.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org




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