Zap Channels hang

[Cliffster]

My system overall works great for the four Metal Shop guys and a couple of SoftPhones, but not good enough to make it to the next step, which is company-wide use for about 25 of us in the near future and another 10 in the next six months. The problem I am trying to solve is a zap channel hang.
I have a TDM410 but I am only using the first two ports now. My incoming ports are connected to two lines on another PBX. The problem occurs when a user calls in, they get to the IVR, and then sometimes route back to the operator. The operator takes a message and both parties hang up, but the zap channels are still active. You can see this using the Flash Operator Panel. All you have to do to correct it is double-click on one of them and the line clears.
Can I set something similar to the silencethreshold used in voice mail to disconnect?
PBX in a Flash Version 1.3 Daemon Status
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* Asterisk * ONLINE * Zaptel * ONLINE * MySQL * ONLINE *
* SSH * ONLINE * Apache * ONLINE * Iptables * ONLINE *
* Fail2ban * ONLINE * IP Connect* ONLINE * Ip6tables * OFFLINE *
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* Ethernet0 * ONLINE * Ethernet1 * N/A * Wlan0 * N/A *
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* Running Asterisk Version : Asterisk 1.4.21.2
* Asterisk Source Version : 1.4.21.2
* Zaptel Source Version : 1.4.12.1
* Libpri Source Version : 1.4.7
* Addons Source Version : 1.4.7
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CentOS release 5.2 (Final) :32 Bit Kernel: 2.6.18-92.1.6.el5

 

[Cliffster]

I think I have figured out that this problem is caused by a parameter on the PBX that this PIAF is tied into. The parameter I need to set on this line on the main PBX is trunk mode = disconnect supervision. Trunk mode allows you to specify either disconnect supervision or unsupervised. Disconnect supervision, also referred to as loop supervision, releases an external line when
an open switch interval (OSI) is detected during a call on that
line. This prevents the line from remaining unavailable for other users. The main PBX is only configured by going through menus on a phone. Changes occur real time and can drop calls, so I am not doing this now. For the time being I just don't route the calls to the main PBX receptionist.