Summary:ASTERISK-08410: Active channels not cleaned upon entering an UNREACHABLE state
Reporter:frawd (frawd)Labels:
Date Opened:2006-12-21 10:47:03.000-0600Date Closed:2011-06-07 14:07:50
Versions:Frequency of
Description:If a SIP device has active bridged calls when it becomes UNREACHABLE (unplugging the phone), the calls are not hanged up.

It can block ZAP lines and prevent others from using it.

The issue happens at least on asterisk 1.2.X (i tried until 1.2.13)


It can be worked around using rtptimeout in sip.conf and putting a ring timeout in Dial commands, but this prevents using Voice Activity Detection (VAD), and can break configurations that use ring timeout = 0 to prevent reaching voicemail for example.
Comments:By: Serge Vecher (serge-v) 2006-12-26 15:31:13.000-0600

using rtptimeout is not a workaround in this case, but a proper way of achieving what you want. I do not believe VAD is currently supported in Asterisk anyway. Can you please elaborate how would this setting affect configuration with ringtimeout?

By: frawd (frawd) 2006-12-28 06:26:31.000-0600

I had issues with a SIP phone becoming UNREACHABLE for some network problems while a call was ringing. Without ringtimeout, the ZAP line would be unusable until Asterisk was restarted.

I use ringtimeout for voicemail or call forwarding like so:
exten => <pattern>,1,Dial(ZAP/1|60)
exten => <pattern>,2,<action on timeout>

To deactivate the action on priority 2, i just remove the ringtimeout, waiting for the phone to hangup like so:
exten => <pattern>,1,Dial(ZAP/1)
exten => <pattern>,2,<action on timeout>

My problem then forces me to use ringtimeout, "breaking" that configuration. I know there are lots of other (and much better) ways to do that kind of configuration using for example the DIALSTATUS variable. Personally, I don't want to wait 60 seconds to have the line available every time a similar problem happens (even if it's not very common). Also anyone who just don't want to use ringtimeout for whatever reason has a problem here.

I just thought it would just be logical to hangup active calls when a phone becomes UNREACHABLE, even if VAD is not supported in Asterisk (although it might be in the future). Otherwise why would rtptimeout be disabled by default?

By: Olle Johansson (oej) 2006-12-31 04:04:04.000-0600

It's an interesting proposal, but has to be made into a configurable option.

The proper way forward is to implement the sip "timer" extension.

This is a feature request and will be closed in a few days unless we get a patch.

By: Joshua C. Colp (jcolp) 2007-01-10 22:59:48.000-0600

There has been no activity on this, no patch, or even a mention that someone would put in the effort. If it is still something you would like done you might try getting people interested by posting to the mailing lists, IRC, etc. Until then though I am suspending this bug. Peace!