From time to time sip channels get locked up and the system keeps the channels alive forever. The same thing happens also on legacy phone calls so on our pstn system we have a paramter called “CallTimeout” to disconnect “runaway” calls.
Is there a way to do this (or something similar) in FreePBX?
If the handset configuration is good then look at RTP Timers on the Asterisk SIP Settings page.
(rtptimeout) = 30
(rtpholdtimeout) = 300
I happened again. The SIP channel stayed open 19 hours. There has got to be a solution for this. I can manually hang up the call with
"channel request hangup SIP/AS5300-00000fed" but I shouldn’t have to do that manually.
Caller ID: 141612341234
Caller ID Name: (N/A)
DNID Digits: (N/A)
State: Up (6)
NativeFormats: 0x100 (g729)
WriteFormat: 0x100 (g729)
ReadFormat: 0x100 (g729)
1st File Descriptor: 647
Frames in: 215003
Frames out: 0
Time to Hangup: 0
Elapsed Time: 19h12m32s
Direct Bridge: SIP/5143690948-00000fec
Indirect Bridge: SIP/5143690948-00000fec
I would look and see if your SIP trunk or device is sending the bye packet. Everytime I have looked into hung channels like this before it was because the carrier was not sending a bye packet. Better to fix the root problem than cover it up.
I appreciate your comment and YES that would be the preferred solution but what if my provider will not (or can not) fix the issue and I can not switch providers because we have a long term contract?
Is there really no way in Asterisk 1.8 to define a channel timeout (hangup) value?