Handset Status Busy When It Is Not


The following is a call progress on a 1.4.6 system. Please note that it shows the handset as state 16 (busy) when it is not. Rebooting the server or restarting asterisk [restart now] solves the problem. I have seen other posts where others are having the same problem, and they were directed to post here. But I have not seen any solutions, and our business is crippled until this problem is fixed.

[code:1] dialparties.agi: priority is 1
dialparties.agi: Caller ID name is ‘Kyle PC’ number is '407’
dialparties.agi: Methodology of ring is ‘ringall’
– dialparties.agi: Added extension 310 to extension map
– dialparties.agi: Added extension 406 to extension map
– dialparties.agi: Extension 310 cf is disabled
– dialparties.agi: Extension 406 cf is disabled
– dialparties.agi: Extension 310 do not disturb is disabled
– dialparties.agi: Extension 406 do not disturb is disabled
dialparties.agi: Extension 310 has ExtensionState: 0
– dialparties.agi: Checking CW and CFB status for extension 310
– dialparties.agi: dbset CALLTRACE/310 to 407
dialparties.agi: Extension 406 has ExtensionState: 16
– dialparties.agi: Checking CW and CFB status for extension 406
dialparties.agi: Extension 406 is not available to be called
dialparties.agi: Extension 406 has call waiting disabled[/code:1]

All of our handsets are Grandstream GXP-2000 with the latest firmware. Please also note that the ExtensionState: 16 is not valid from what I have read.


I forgot to mention how to cause this to happen. The easy way to create this error is to simulate an incoming call by dialing 7777 (in this case from extension 407), answering at an extension (in this example 406), transferring the call to another extension (in this case 310), then terminating the call. Any succeeding calls from internal or external will find extension 406 as busy, when it is not.

Thanks, Kyle

This is the same bug as here:

Setting call-limit=4 (or some other none-zero value) in each SIP extension appears to fix it, but at the moment you have to manually edit the file after doing any changes in FreePBX.

please provide some testing on Asterisk version 1.2 with the call-limits set so we can get feedback if this breaks anything on 1.2 or not? Then we can go forward and add this option.


I am running Asterisk 1.4.6. Please forgive my ignorance, but are you asking me to change to 1.2.x of Asterisk? Or are you asking me which version of freepbx I am running? I have been unable to figure out how to find the version of freepbx, and always assumed it was the same as the Asterisk version.