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eherr
eherr's picture
Joined: 2011/11/11
Points: 0

So with the latest MTE downloadable version, I was/am having an issue when the far end Polycom Phone or Sip device loses network connectivity, the MTE Asterisk will still play audible ring tones as if the phone is actually ringing.

Test extension has qualify = yes. I believe that defaults to 2000ms if I am not mistaken; been awhile since I looked that up.

I know the extensions technically register to Kamailio.

When I run kamcmd ul.dump, or more specifically, kamcmd ul.lookup location account, there is a field for Expires.

I found out through testing, this is populated from the sip device; ie.) registration timeout variable on the Polycom Phone or Soft phone, etc.

I typically set this field to 300 seconds.

I know/if I recall correctly on the older Thirdlane installations, the SIP Invite was being sent to the end device every 2000ms and updating the "OK|UNKNOWN|LAGGED" and based on receiving, or not, the 200 OK response back.

Basically want the phones to become unregistered/unknown like they used to so I dont get false ringtones to the callers. This is an issue for hunting based extensions as it wont roll over and we could potentially miss calls for up to 300 seconds.

Is there any way to get this condition back where phones would become unknown after a short time or is this now frowned upon in the community?

What are my options?

Thanks in advanced

volodya
volodya's picture
Joined: 2017/01/05
Points: 280

Hello,

Thank you for the suggestion. We already did some work to support this and will be adding it to the new release.

For now, you could make the hunt list timeout shorter. Is it an option?

eherr
eherr's picture
Joined: 2011/11/11
Points: 0

For some applications but not all applications.

Is this going to be release in a year or sooner with an update package?

Any mods I can make with thirdlane or kamailio at this point?

I believe the shortest Expires value with Kamailio is 60 seconds.

eherr
eherr's picture
Joined: 2011/11/11
Points: 0

also to clarify for future readers, I believe it is a SIP method OPTIONS, not INVITE, query sent to the sip device.

volodya
volodya's picture
Joined: 2017/01/05
Points: 280

Hello,

You should reduce the hunt list ring timeout to a smaller value and have an appropriate "destination on no-answer" setting to make sure that calls are handled efficiently. https://www.thirdlane.com/docs/platform/hunt-lists#extension-based-ring-groups

By the way, what is the current value of the "TL_RINGTIME" variable https://www.thirdlane.com/docs/platform/dialplan-variables? Did you consider using Device Based Ring Groups or Queues?

Also, did you investigate if and why phones are not receiving requests? It is possible that the router is destroying sessions on the timeout and the qualify frequency should be increased.