Re: [asterisk-users] gxp2000 expansion module blf leds not working
Am I the only one using the GXP2000 expansion module? Thanks, Zoilo. Zoilo Gomez wrote: Today a 56-button expansion module for the GXP2000 came in. When I program the buttons+leds on the expansion module for BLF, then speed-dial works fine: when I press the button the programmed ext number is called properly. However the LEDs are always off: neither green nor red They are not broken, because on reboot the LEDs flash red! On the GXP2000 itself, this function works fine, with LEDs being green when the ext is free, or red whenever it is busy. Does anybody know this problem? Or can anyone confirm that the LEDs on the GXP2000 expansion module should be working properly? Thanks, Z. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gxp2000 expansion module blf leds not working
Today a 56-button expansion module for the GXP2000 came in. When I program the buttons+leds on the expansion module for BLF, then speed-dial works fine: when I press the button the programmed ext number is called properly. However the LEDs are always off: neither green nor red They are not broken, because on reboot the LEDs flash red! On the GXP2000 itself, this function works fine, with LEDs being green when the ext is free, or red whenever it is busy. Does anybody know this problem? Or can anyone confirm that the LEDs on the GXP2000 expansion module should be working properly? Thanks, Z. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] b410p + fax (echo cancellation)
I really don't get it From several emails in this list archive, I had clearly understood that it is important to switch Echo Cancellation off for fax-channels, or faxing would not work properly. However, faxing (B410P ISDN bridged to TE410P PSTN) seems to work fine with EC at 256 taps on the B410P. I am confused; can anyone enlighten me? Thank you! Z. == Zoilo Gomez wrote: We have recently purchased a B410P Digium 4* ISDN-2 card with hardware EC. On the same server, I also have a regular Digium 4-channel PSTN-card (TDM410P ?), used to interface to some analog devices, a.o. 2 fax machines. For faxing, EC needs to be off (or so I understand from the archives). How can I switch EC off for an ISDN B-channel if a fax is coming in? Z. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] b410p + fax (echo cancellation)
We have recently purchased a B410P Digium 4* ISDN-2 card with hardware EC. On the same server, I also have a regular Digium 4-channel PSTN-card (TDM410P ?), used to interface to some analog devices, a.o. 2 fax machines. For faxing, EC needs to be off (or so I understand from the archives). How can I switch EC off for an ISDN B-channel if a fax is coming in? Z. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP + short numbers + name of customer
We are using a couple of Grandstream GXP2000 SIP-phones with Asterisk. In our dial-plan, we have implemented a list of short numbers in extensions.conf, like: exten = 1234,1,Dial(Zap/0987654321) So when I pickup the SIP-phone, and I dial 1234, the system dials 0987654321 and connects me to that customer. Unfortunately I cannot see the name of the customer, and I do not know if perhaps I punched the wrong short number. Is there a way to have Asterisk print the name of the customer on the SIP-phone display, instead of 1234? Maybe the implementation (see above) is not optimal, and there is better way to deal with these short numbers? The same question for incoming calls: it would be great to have Asterisk print the name of the customer on the display when a call comes in, instead of his phone number 0987654321. Z. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About BRI / ISDN hardware. What to buy?
We are using Sirrix (http://www.sirrix.com). 4* BRI for app. € 550. Works fine; has echo cancellation as well. Cosmin Prund wrote: Hello everyone. I need a BRI ISDN card that works in Romania. I already have one of the Cologne HFC-S PCI cards and it doesn't work right, it's junk. I get wy too much echo using it. I'm now shopping for a better card. Can anyone recommend me a card that fits the following: (a) Costs less then $1000 / 750 euro (b) Has one or (preferably) two ISDN S0 interfaces. (c) Easy to set up. (d) Drivers offer proper echo-canceling OR has an hardware echo canceler. I might increase the $1000 a bit if I can get good hardware echo canceler... Thanks, Cosmin Prund ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P rings randomly when phone line makes call
Yuan LIU wrote: Not sure if anyone experienced the same - or if anyone ever connected a POTS phone to the Phone jack on an X100P card. The POTS phone rings normally when the FXO receives a call. The POTS phone can also make outgoing calls when FXO is not holding the line. This is desired. But if a call connected to the POTS phone lasts longer than a couple of minutes, Asterisk would receive ring conditions from X100P at seemingly random intervals, and kick off incoming dialplan associated with the Zap channel. I experienced a similar problem, when my FXO was not the only port terminating the PSTN-interface, but I had a normal phone set connected in parallel as well. I turned verbose to 6 and debug to 6, but all I could see was: -- Starting simple switch on 'Zap/1-1' Any idea how I can find the cause? Thank you. System is Asterisk 1.2.13 on Ubuntu 6. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
