[asterisk-dev] rtp timeout failover

[Wolfgang Pichler]

Hi,

2010/6/1 Olle E. Johansson <oej at edvina.net>:
>
> 1 jun 2010 kl. 15.45 skrev Wolfgang Pichler:
>>
>> call leg A does dial call leg B over a channel with rtp data. call leg
>> B rtp data does timeout -> so call leg A continues at
>> context/extension/prio...
>>
>
>>
>> I am currently locking at chan_sip from asterisk 1.4.21 - there around
>> line 15725 is the rtptimeout handling - as far as i can see it does
>> simple hang up the call leg where the rtp timeout did occoured - so
>> dialplan execution will continue for call leg A - but it will look
>> like the hangup was a normal hangup. What about adding a new channel
>> variable to call leg A - which does mean that the hangup is because of
>> a rtp timeout ?
> The channel variable would then be set in the outbound channel that we
> will destroy...
If we get the bridged channel -- and set the channel variable to this
channel - then the variable will still be available - because the
bridged channel does continue in the dialplan, or do i see this wrong
?
>>
>> An other possibility would be to create a new HANGUP Cause for this -
>> but i think this change would have many side effects... - so not the
>> best way
> THat may be the only useful way. What cause are you getting today?
> Anything useful?
Normal call clearing... how hard would it be to add an additional
hangup cause ? Or what about a hangup sub cause ? So the main hangup
cause variable can get extended...

>
> /O
>
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