Devices and Users Call Waiting disabled Not Working


I’m currently using Asterisk Ver. 1.2.18 with this great FreePBX Application

The problem I’ve is when Diabling Call Waiting in the User Screen, there is no change made and with my Softphone Zoiper I still get second or third Caller waiting an beeping.

for example:
my device is a SIPdevice 1200 and user is logged on via *11100 password is User 100 with DID 123456789 and call waiting disabled. 1st call comes in > 100 answers > 2nd call comes in and is waiting > 3rd comes in and is waiting and so on instead of getting a busy or a hangup, because voicemail is turned off. Even the Feature Code *71 deactivating Call Waiting doesn’t work.

Any Idea?



I think the Problem is:

-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi

dialparties.agi: Starting New Dialparties.agi
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Manager ‘admin’ logged on from
dialparties.agi: Caller ID name is ‘XXXXXXXXX’ number is 'XXXXXXXXX’
dialparties.agi: USE_CONFIRMATION: 'FALSE’
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 100 to extension map
– dialparties.agi: Extension 100 cf is disabled
– dialparties.agi: Extension 100 do not disturb is disabled
> dialparties.agi: extnum 100 has: cw: 0; hascfb: 0 [] hascfu:
0 []
> dialparties.agi: ExtensionState: -1
dialparties.agi: Extension 100 has ExtensionState: -1
– dialparties.agi: Checking CW and CFB status for extension 100
– dialparties.agi: dbset CALLTRACE/100 to XXXXXXXXX
== Manager ‘admin’ logged off from
– AGI Script dialparties.agi completed, returning 0
– Executing Dial(“SIP/”, “SIP/1200||Ww”) in
new stack
– Called 1200
– SIP/1200-00939de0 is ringing
– Channel 0/22, span 1 got hangup request
– Hungup ‘IAX2/asterisk14-2’

hints, and thus callwaiting don’t work right with adhoc devices. I’ve been making some changes on 2.4 (svn trunk) which you could take a look at. They would work fine on 2.3 also if you want to try and pull the changes to your system. They require changes in the login/out macros and a new agi script that goes with them.

Philippe Lindheimer - FreePBX Project Lead
http// - IRC #freepbx