On Wed, 11 Mar 2009, Rosa De Santis wrote: > > Hello all. > > Please, I'd like to know if somebody can help me with this problem. > I have successfully configured a PBX with Asterisk 1.4 and a Digium analog > card with 4 ports. > > This PBX has a lot of incoming and outgoing calls, and works perfect in > general, but there are some extrange cases where an incoming call is > bridget with an outgoing call, and the caller that is calling TO the PBX > can even hear the dtmf tones of the caller that is calling OUT the PBX, > and due the high traffic this is happening a lot. It seems that asterisk > is taking the zap channel to call out in the exact moment before it is > marked as busy with the incoming call. Please, is there any > configuration to avoid this?
It's called "glare" and your options to "fix" it partly depend on what country you are in. I don't think it's totally fixable with analogue lines. Make sure you have the right country code specified in /etc/zaptel.conf and are using "Kewlstart". Eg. for the UK: fxsks=1 fxsks=2 fxsks=3 fxsks=4 loadzone=uk defaultzone=uk Your options are to replace the 'ks' with 'gs' or 'ls' - but others might be able to advise you what's best for your country/telco. But do make sure your incoming calls start at one end, and outgoing start at the other - so if the first line is connected to port 1, 2nd to port 2, etc. you want to dial-out starting at port 4 - that's the capital G option in dial - eg. dial(Zap/G1/xxxx...) the lower-case g will start outbound dialing at the lower port number. (And put the lines in the right group in /etc/asterisk/zapata.conf) Another solution might be to get more channels - are your users & callers complaining of busy tones? Gordon _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
