[asterisk-users] Call didn't stop after losing one leg
jcolp at digium.com
Tue Jul 15 07:36:17 CDT 2014
> Hello there,
> I'm using a Debian box with Asterisk 22.214.171.124 as a DID-PSTN gateway,
> so I can receive calls in a DID number and redirect it to my mobile line.
> It has been working flawlessly for a few months, but I have noticed
> that some calls were not cut after losing one leg (the one with the
> DID server), and kept the PSTN leg active until I restarted the
> server (with the unexpected cost involved in the PSTN call).
> The relevant extensions.conf line is:
> exten => 34911234567,1,Dial(SIP/pstn/447123456789)
> And both DID and PSTN sip accounts have canreinvite=yes, so they
> can have direct media.
> I haven't collected any debug log nor any other relevant information.
> Does anybody know why something like this happens, or how can I
> cut a call that unexpectedly losed a leg?
Since you are having media go directly the only thing that can be
monitored is the signaling of the call itself. This can be accomplished
using SIP session timers. There is a section "SIP Session-Timers" in the
sip.conf.sample file which has the various configuration options
relating to it.
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