[Asterisk-Users] Potential call logging problem for commercial systems..

[Peter Brown]

At 13:03 14/11/03 -0800, you wrote:
>>I have been playing around a lot with the CDR today and I may have 
>>stumbled across a very serious problem, specifically where there is 
>>billing taking place..
>>
>>If a call is placed between 2 phones and the network connection is broken 
>>from both the phones with out hanging up first the call is never logged 
>>to the CDR and it seems never termintaed.. It would appear that Asterisk 
>>relys on recieving the SIP signals to tell it that the call has 
>>terminated and so if it does not get that data it will never release the 
>>call, there does not appear to be any call progress checking that could 
>>terminate the call if the end points could no longer be reached..
>>
>>This could have a major impact on, not only billing services, but also on 
>>things like IAX trunks that could sit with an open channel forever..
>>
>>I may have missed something in the config that would solve this problem 
>>so if I have please reply and let us know what it is..
>>
>>Later..
>
>You should set a maximum timeout on any channel you bring up.  This kills 
>SIP zombies.  It's not optimal, but prevents infinite calls.
>
>Alternately, if your RTP data stream is passing through Asterisk during 
>the call, you could write a short extension to chan_sip which would look 
>for "imbalanced" traffic or zero traffic.  In other words, if one side of 
>the conversation kept sending audio data, but the other side became 
>completely silent, then after a few minutes (configurable timer?) you 
>could probably assume that the other side was disconnected.

Boy, there would be a lot of girlfriends really annoyed by their beaus 
hanging up on them!!

>Same thing if both side of the RTP stream were happily sending data, and 
>suddenly they both stopped; kill the call.
>
>See http://bugs.digium.com/bug_view_page.php?bug_id=0000207
>
>JT
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