[VoiceOps] SIP calls that aren't torn down

Darren Schreiber d at d-man.org
Tue Jun 7 15:56:55 EDT 2011

Is your softswitch from 1982? ;-)

OK OK I kid...  I am seriously curious what you're using though... These
are pretty standard in the open-source world and also on phones and
ATAs/media gateway hardware, etc.

Maybe you just aren't looking for the right option? Look for anything you
can configure known as a "timer" and kick back what you've got...

- Darren


On 6/7/11 12:46 PM, "Frank Bulk" <frnkblk at iname.com> wrote:

>That feature is not in our softswitch.  Sounds like a feature request I
>to make.
>-----Original Message-----
>From: Darren Schreiber [mailto:d at d-man.org]
>Sent: Tuesday, June 07, 2011 2:29 PM
>To: frnkblk at iname.com; VoiceOps at voiceops.org
>Subject: Re: [VoiceOps] SIP calls that aren't torn down
>This feature (session timers or RTP timers) is usually built into your
>switch, are you stating that you're not utilizing such a feature?
>- Darren
>On 6/7/11 12:25 PM, "Frank Bulk" <frnkblk at iname.com> wrote:
>>Over the last few months our softswitch has accumulated 10 "stuck" calls
>>where there's no media traffic.  From what the softswitch vendor can tell
>>guess, it just didn't receive a BYE to tear down the call.
>>We know that most SIP traffic on 5060 is UDP, and UDP is connectionless.
>>How are other vendors and systems managing such scenarios?  I suggested a
>>"no media" test where after x hours of no media, to tear down the call
>>log the info.
>>VoiceOps mailing list
>>VoiceOps at voiceops.org

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