[asterisk-dev] Media TimeOut for SIP and IAX Trunk

bilal ghayyad bilmar_gh at yahoo.com
Thu Jun 5 14:49:34 CDT 2008


Dear All;

Big thanks for the kind help and nice answers I got.

But I have one case need to know if it the hangup will
be done successfully: 

If there is an IAX trunk between two Asterisk
machines, and if the call was initiated by the first
Asterisk Box to the second Asterisk box via that IAS
trunk, and if the Internet connection with the first
Asterisk become down (disconnected), in that case:

Will the hangup will be applied on the IAX at the
other Asterisk Box (destination), and if the call was
implemented via the Zap Channel, then the Zap Channel
will be free as the hangup will close the call?

Your kindly help is highly appreciated.

Regards
Bilal




-----------------------
On Jun 4, 2008, at 2:51 PM, Tim Panton wrote:

>
> On 4 Jun 2008, at 17:12, bilal ghayyad wrote:
>
>> Hi List;
>>
>> Any one can advise me if IAX trunk support media
time
>> out to disconnec the call automatically after
certain
>> vaule of timeout in case no more media running
between
>> end points and hangup signal was not available
(which
>> happens in alot of cases)?
>>
>> And in case of SIP, I heared it is existed but I
have
>> to check the session timer in the trunk version of
the
>> sip_chan, but really I do not know how to check
this
>> and wether there is any need to be done on the
source
>> to be modified and compiled to obtain it
successfully
>> working.
>>
>> Any help?
>>
>
> Isn't this is the default behavior in IAX2? Normally
IAX
> doesn't separate the media from the control, so
generally
> IAX will drop a call a few seconds after the last
received packet.


That is absolutely correct.  The call will be dropped
once the first  
full frame is not acknowledged after some period of
time.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.






      



More information about the asterisk-dev mailing list