[Asterisk-Users] Re: DSL (DMT) goes down when X100 plugged in
My guess was the 100 presented too low an impedence to the line. So, I took an answering machine that had a phone jack on it (pass-through). I plugged the ans. machine into the filter and the 100 into the ans. machine. Everything works now. I can also try a second filter. Thanks, Tom Schaefer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DevLite problem with ztcfg
Hello all, I finally got around to installing my Dev Kit Lite. I did the install yesterday from the latest CVS. I am receiving an error that does not let * start up. When I go through the procedure to load the modules, I get the following error after running ztcfg. Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? I looked in the archives but could not find a valid solution. The config file is: fxoks=1 fxsks=2 loadzone=us defaultzone=us When I did this with only the 100 card, it did work, but that was on a CVS release from last week. I ran the astinstall script and un-tarred the configs into / Any help would be appreciated. Thanks, Tom Schaefer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More external call control
I have some questions for anyone that can help. I discovered an email in the archives about someone adding an external call control router on WIndows 2003, but could not find a reference to the code. I wanted to see how far I could go with AGI scripts before having to modify the code. I have been looking at AGI scripts and I think I have some of the answers to this, but here goes... What I need to do is the following: 1. Call comes into a pilot number. 2. A message is sent to a script indicating uniqueid has been queued. Included with this info is the calls profile information like DNIS, ANI, etc. 3. After some period of time, a script sends a message back to * to have the call (by uniqueId) routed to a specific extension on the switch. 4. When the caller or agent hangup, a message is sent to a script. My questions are as follows, (but before I begin; I know there is queueing and some ACD functionality in *, but I need to do this externally. I want the queueing decisions to be external because my central queue engine handles things like email, chat, etc as well as calls): In other words, can I send some message to * that will tell it to route a call in queue to a specific extension by a unique ID (because there may be los of calls queued). While the call is in queue, can I send commands to have different announcements played? If a call hangs up while in queue, is that a step in extensions.conf so I can call my script with that info? Thanks, Tom Schaefer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Smallest server continued...
This thread got me thinking of other servers that would run asterisk. The obvious question comes up if Xebian (the xbox version of Debian) would run as a SIP only server? Asterisk on an XBox would be a small box! Cheap too. Tom Schaefer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
