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Sip Registrations Offline Not Updating

Posted by eherr on Mon, 01/04/2021

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


Submitted by volodya on Mon, 01/04/2021 Permalink

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?

Submitted by eherr on Mon, 01/04/2021 Permalink

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.

Submitted by eherr on Mon, 01/04/2021 Permalink

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

Submitted by volodya on Tue, 01/05/2021 Permalink

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.

Submitted by eherr on Wed, 04/21/2021 Permalink

To further this a bit more, the issue is remote users.

The following is what appears to be the condition.

* Perfect World
- Kamailio sends a qualify options to end point. Endpoint sends back 200 OK. ( Happy )
- endpoint is tabled in userloc and has a timeout of 300seconds.

* Poor Network
- userloc just received a registration update provided by the endpoint, value = 300
- Endpoint goes offline.
- qualify message goes out during normal interval, doesnt receive a response back from endpoint, tries 10 more times before giving up.
- userloc still show endpoint tabled and timeout is now say 230 seconds
- you call the extension, the caller hears ringtones even though the endpoint is down. this continues for 230 seconds until the userloc expires.

this was handled in previous versions sans kamailio by the lack of the qualify response making the phone unreachable and the calling party would go to voicemail or jump to the next iteration in the hunt group.

thoughts?

In reply to by volodya