[S]WebPhone On Hook Periodic Ringing at 15,20 mint Intervals

[streetboy]

I have set the phone On Hook=Y, and there is a call coming in every 15, 20 mints. Even when there is no real call. It's like an asterisk just calling the phone periodically. Is there any way we can increase that time or totally disable it? And secondly, can we change that periodic ring sound to actually be different from the inbound call ring?


VERSION: 2.14-897a
BUILD: 230927-0857
Asterisk 16.17.0
Version: 2.14b0.5
SVN Version: 3769
DB Schema Version: 1697
Scratch Install - Alma 8

[carpenox]

Are you seeing this activity in Asterisk? That doesn't make sense

[streetboy]

Yes. It is a simple call.

Code: Select all

 == Manager 'sendcron' logged on from 127.0.0.1
  == Using SIP RTP CoS mark 5
    -- Called 8001
    -- SIP/8001-00000bc7 is ringing
  == Manager 'sendcron' logged on from 127.0.0.1
  == Manager 'sendcron' logged off from 127.0.0.1

Auto Answer Webphone = NO.

We hear this ring every time the agent logs in, and the ring will stop after 10,15 seconds, and the webphone shows up as Registered, if that call is answered(Manually) there will be a lady saying "You are the only person in the conference", but we do not want that otherwise the actual inbound call will not be routed to that agent, because that SIP account is already busy. Why we have set it up like this because the client doesn't want the calls to be auto-routed to the closers. They want the closer to have the ability to ring their phones when they get that transfer and they will accept the inbound call manually after clicking on the phone symbol on the webphone interface. So now they are getting these periodic calls from Asterisk. What I believe is that it's like a calibration call initiated from an asterisk to make sure that the phone is logged in/registered. If that is the case I am looking for a setting to increase that increase periodic call if it's not possible to disable it completely and change the ringing sound so we can differentiate it from the actual inbound calls.

[carpenox]

Check in sip.conf for rtpkeepalive and uncomment it and set it, also raise the timeout for rtp

[streetboy]

Thank you for pointing me in the right direction, life saver as always.
Tweaking these helped out.
maxexpiry= ; Maximum allowed time of incoming registrations
minexpiry= ; Minimum length of registrations/subscriptions (default 60)
defaultexpiry= ; Default length of incoming/outgoing registration

[carpenox]

Glad it worked out