2 thoughts on - Hangup Chanel When A Peer Unregisters

• rtptimeout= in sip.conf will hangup a channel if no rtp is received for a period of time.

• From: asterisk-users-bounces@lists.digium.com
  [mailto:asterisk-users-bounces@lists.digium.com] Hello group and thank you for the attention.

  I’m using Asterisk 11.12 running on Ubuntu Server 12.04

  We have an issue with channels remaining open after a SIP peer unregisters.

  It seems that if the peer goes away before manually hanging up a call, the channel remains open until a hangup request is sent from the CLI.

  Is there any way to drop a channel when the peer using it disappears?

  Googled every phrase I could think of. No luck.

  Thank you!

  Pat Collins

  rtptimeout= in sip.conf will hangup a channel if no rtp is received for a period of time.

  Thanks for the response Gareth.

  The problem is that I may have a conference call up for days at a time.

  During this time, there may be no activity for hours.

  If the endpoint the endpoint is able to send RTP keepalive packets, your solution is spot on.

  Will have a look at it.

  Thanks again!

  PC…