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lm at redabierta.es
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 Posted: Sun Jul 13, 2014 3:28 am Post subject: [asterisk-users] Call didn't stop after losing one leg |
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Hello there,
I'm using a Debian box with Asterisk 1.8.13.1 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?
Thanks!
L.
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jcolp at digium.com
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| Posted: Tue Jul 15, 2014 7:36 am Post subject: [asterisk-users] Call didn't stop after losing one leg |
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lm wrote:
Kia ora,
| Quote: | I'm using a Debian box with Asterisk 1.8.13.1 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.
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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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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lm at redabierta.es
Guest
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| Posted: Tue Jul 15, 2014 8:23 am Post subject: [asterisk-users] Call didn't stop after losing one leg |
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Hi Joshua,
| Quote: | 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.
|
I have been reading about it and it looks like an effective way to
mitigate this problem. I'll test it and see how it goes with my providers.
Thank you!!
--
_____________________________________________________________________
-- 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 |
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