[asterisk-dev] channel stay up when extension unreachable

[Anton Raharja]

On 08/24/2010 05:21 PM, Stefan Schmidt wrote:
> Anton Raharja schrieb:
>   
>> 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
>>
>>
>>   
>>     
> hello,
>
> you could also use totaltimeout but this will also disconnect a call 
> which is still active.
> But normally a dahdi channel will not run forever, cause the 
> service/person on the other side will disconnect the call after some 
> time of no activity.
>
> maybe you could use a totaltimeout of 2 hours or some value you think 
> you will normally not reach.
>
> best regards
>
> steve
>   
Hi,

Will try that, thanks.

Regarding DAHDI stay up, unfortunately I have some example of DAHDI
channel stay up in cases I asked, our CDR recorded conversations as long
as 8 hours from either SIP to DAHDI or fxs to DAHDI fxo which was
impossible. I guess the rtptimeout option will solve the problem on SIP
to DAHDI fxo case.

At that time we used 1.4.32 and 33.1

When using DAHDI fxs, If I'm not mistaken, it was involving transfer
calls and before answered the extension making the transfer got
unreachable, and leave the channel up, also making DAHDI fxo port
unusable until soft hangup, or restart.

anton