All, Experiencing some issues on my FXO lines. If a call comes in on an FXO and then get transferred to another FXO (say to call someones cell phone), those two lines will stay tied together indefinitely. This happens to us when we transfer an incoming call to our on call guys after hours and on weekends. We have installed 3 other * boxes and they do the same thing. We use a Adit Channel bank for all incoming FXO and the other installs use multiple 1 port digium FXO cards or a combination of the 1 port and 4 port FXO cards. Has anyone else experienced this and if so how did you fix it? Thanks in advance....................>Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com <http://www.golden-tech.com/> 219-462-7200 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040618/3a1f2711/attachment.htm
> Experiencing some issues on my FXO lines. If a call comes in on an FXO and then gettransferred to another FXO (say to call someones cell> phone), those two lines will stay tied together indefinitely. This happens to us when wetransfer an incoming call to our on call guys after> hours and on weekends. We have installed 3 other * boxes and they do the same thing. > > We use a Adit Channel bank for all incoming FXO and the other installs use multiple 1 portdigium FXO cards or a combination of the 1 port> and 4 port FXO cards. > > Has anyone else experienced this and if so how did you fix it?If you're using plan old pstn analog lines, put a voltmeter on the analog line to see if you have call supervision coming from the telco. You should see the voltmeter either dip to zero volts for about a half second, or, voltage reverses for some short period of time. That should occur within a few seconds after the caller hangs up. Exactly "when" that occurs varys by central office switch manufacturer. If you don't see any form of supervision, then you have to implement some sort of timer, tone detection, etc. You can also talk to your telco engineering/tech folks to see they have any options for call supervision. (The sales office won't have a clue in most cases.) Other choices might include changing your pstn lines from loop start to E&M signaling, etc.
Thanks for the advice, checked it with a VOM and sure enough, it had remote disconnect supervision coming down the line, but we didn't have the dip switches on the FXO card on the ADIT switched correctly. Once we did that all is good. Thanks again.........>G Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Rich Adamson Sent: Friday, June 18, 2004 9:31 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FXO Issues> Experiencing some issues on my FXO lines. If a call comes in on an FXOand then get transferred to another FXO (say to call someones cell> phone), those two lines will stay tied together indefinitely. Thishappens to us when we transfer an incoming call to our on call guys after> hours and on weekends. We have installed 3 other * boxes and they dothe same thing.> > We use a Adit Channel bank for all incoming FXO and the other installsuse multiple 1 port digium FXO cards or a combination of the 1 port> and 4 port FXO cards. > > Has anyone else experienced this and if so how did you fix it?If you're using plan old pstn analog lines, put a voltmeter on the analog line to see if you have call supervision coming from the telco. You should see the voltmeter either dip to zero volts for about a half second, or, voltage reverses for some short period of time. That should occur within a few seconds after the caller hangs up. Exactly "when" that occurs varys by central office switch manufacturer. If you don't see any form of supervision, then you have to implement some sort of timer, tone detection, etc. You can also talk to your telco engineering/tech folks to see they have any options for call supervision. (The sales office won't have a clue in most cases.) Other choices might include changing your pstn lines from loop start to E&M signaling, etc. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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