Dealing with a sip phone that's been left off the hook?

jayrod422
Joined: Thu 12 of Oct, 2006

Re: Dealing with a sip phone that's been left off the hook?

Posted:Tue 17 of Feb, 2009 (20:24 UTC)
check out rtptimeout

if the phone isnt transmitting any audio asterisk will hang it up

contact me off list if you want jayrod422 at fed-com.com
RobRobBob
Joined: Sat 27 of May, 2006

Dealing with a sip phone that's been left off the hook?

Posted:Sun 08 of Feb, 2009 (00:33 UTC)
Hi there,

I have a small asterisk setup in my house. I have one grandstream ATA (286 HT) set up as a SIP client for all incoming and outcoming calls for my voip calls. I noticed recently, if one of the analog phones connected to the ATA is left off of the hook, it will automatically pick up an incoming call. This isn't good, because if someone calls the house, they'll get a phone with no one on it! :)

I would like my set up handle this better, perhaps to instead start beeping or hang up the connection? Is there a standard way to set this up? Is this usually configured in the extensions.conf file, or on the ATA? I have looked through the info here for tips on how to deal with this, but haven't found anything to solve my problem.