Rosa De Santis
2009-Mar-11 16:38 UTC
[asterisk-users] Problem with incoming and outgoing calls via TDM
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? Thanks a lot in advance. Rosa. _________________________________________________________________ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090311/0ddaf479/attachment.htm
Dave Fullerton
2009-Mar-11 16:58 UTC
[asterisk-users] Problem with incoming and outgoing calls via TDM
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? > > Thanks a lot in advance. > Rosa.The situation you're referring to is called glare. You'll find discussion of it in the archives and on voip-info.org. You need to make sure you are seizing lines for outgoing calls in the reverse order that they are used for incoming calls. Check out the G dialing option for Zaptel/DAHDI channels (under Dialing a Group section): http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels If this doesn't work, your next best bet is to increase the number of lines you have. -Dave
Gordon Henderson
2009-Mar-11 17:26 UTC
[asterisk-users] Problem with incoming and outgoing calls via TDM
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