[asterisk-dev] channel stay up when extension unreachable

[Anton Raharja]

On 08/24/2010 04:24 PM, Stefan Schmidt wrote:
> Anton Raharja schrieb:
>   
>> === electricity down in 801's room and 801 became unreachable:
>>
>> [Aug 20 14:46:45] NOTICE[8052] chan_sip.c: Peer '801' is now
>> UNREACHABLE!  Last qualify: 7
>>
>>   
>>     
> this is the normal behavior, cause even if the options packet coudn´t 
> reach the phone, the rtp session may still be up.
> You could use rtptimeout to prevent such events so the call will hangup 
> automagical if the rtp session dies without a bye.
>
> best regards
>
> steve
>
>   
Hi,

yup, not a bug at all, after rtptimeout=30 I got this:
Disconnecting call 'SIP/888-00000000' for lack of RTP activity in 31 seconds

thx

One more question, if this is also normal and rtptimeout will resolve it:
case 1: a call made from SIP to DAHDI and channel stay up even after SIP
unrechable
case 2: a call made from DAHDI fxs to DAHDI fxo and channel even after
DAHDI fxs hangup by caller mistake

anton