Re: [Asterisk-Users] Can One FXO Support Multiple Phone Lines?
More then likely he used 4 conductor (two pair) cable and meant 2 lines (phone numbers) per cable. They will need to be broke out to separate jacks. --JR - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 9:10 PM Subject: RE: [Asterisk-Users] Can One FXO Support Multiple Phone Lines? Definitely one line per FXO port, but the wording of the original poster was two numbers, not two lines, and while it may not be universally true, distinctive ring should allow two (or more) phone numbers to be present on an FXO port, and asterisk should be able to tell which one is calling. If the original poster did mean lines and not numbers, maybe there was some confusion about the difference between PSTN and ISDN. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Tuesday, 14 March 2006 12:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Can One FXO Support Multiple Phone Lines? In a word...No. One line per FXO port. Next question? Michael --Original Message Text--- From: Andrew Berman Date: Mon, 13 Mar 2006 19:32:29 -0500 I am currently having our new office wired up with 8 PSTN lines. The guy asked me if he could wire it up such that one line had two phone numbers. I bought a Sangoma A200 with 8 FXO ports, but now I'm wondering if all I needed were 4 FXO ports. Is it possible to set up Asterisk with 2 numbers per FXO? Thanks for any help, Andrew -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --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] OT O'Reilly Asterisk TFOT
Does the printed version have an index? -- JR Whilst it can be downloaded I find that a paper copy is easier to read. I bought it for that reason alone. I also find it's a usefull addition to my tool box. I can't always access the net whilst on site. If I get stuck doing something I can look it up in the book. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dave Cotton wrote: I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. ___ --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] Re: Gain adjustment
On Tue, Jan 31, 2006 at 08:44:18AM -0600, Rich Adamson wrote: When adjusting the rxgain and txgain in Asterisk 1.2.1 do I need to restart Asterisk or is it enough to just reload Asterisk in order to apply the new setting? Need to stop asterisk and restart it. A reload will not take the new setting into consideration. There is no need to stop/start the zaptel drivers, just asterisk itself. Actually there is a restart in chan_zap of Asterisk 1.2. And zapata.conf is of chan_zap. However I'm not sure exactly how much of those changes do apply on repload. Anyone know if the gain can be adjusted on a channel basis? I have a TDM412B card (1-FSX, 2-FXO) and adjusting the gain in zapata.conf affects all three. Is there a different syntax? This is a problem for my as my FSX need no gain and doing so causes problems and my need a gain set to about 15 else you can't barely hear the caller and VM is worse. Thanks JR. ___ --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] Re: Dell PowerConnect 2724 Switch and QoS for VOIP?
- Original Message - From: Brian Roy [EMAIL PROTECTED] Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Friday, January 20, 2006 6:12 PM Subject: Re: Dell PowerConnect 2724 Switch and QoS for VOIP? On 1/20/06, Sean Tempesta [EMAIL PROTECTED] wrote: Has anyone had any experience with the Dell PowerConnect 2724 Gigabit Switch and VoIP traffic? It seems to support Class of Service (CoS), but not full Quality of Service (QoS). Honoring, and tagging, are completely different things. I believe that the 2724 will just honor QOS frames. The 6000 series Dell switches are the only true Layer 3 switches with QOS. You will probably get less than stellar reviews from this crowd but it's not a bad switch. I've used a 6024 as a building core switch and not had a lick of problems. -Brian Brian, I'm new to this QoS stuff but can't QoS be set at the device? At least it's an option on the Grandstream GXP 2000 Layer 2 (Default Blank) and Layer 3 (Default 48). JR ___ --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] Need HT488 FXO example for both inboundandoutbound.
Bjorn, Thanks!! The example looks like what I need although I don't understand the Forward to VoIP as I don't have an ITSP. I'll give it a try later today. Basic Settings: Number of rings: 1;0 is not a valid option Forward to VoIP: a number in your from-pstn context where you want to receive incoming calls Thanks again and happy new year... JR - Original Message - From: Bjorn Asmul [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 01, 2006 11:55 AM Subject: RE: [Asterisk-Users] Need HT488 FXO example for both inboundandoutbound. Hi James, This link might help: https://billing.atlasvoice.com/forum/index.php?topic=20.0 -- Bjorn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Ronald Sent: Sunday, January 01, 2006 1:56 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Need HT488 FXO example for both inbound andoutbound. I'm new to Asterisk and I'm looking for example of how to set up the FXO side of an HT488. I have the FXS side working and can place calls between it and soft phone just fine. What I was able to find the Wiki, forums google has not been useful to me. I think I'm missing something simple probably on the HT488 device. Once I have working example I'd be happy to post it on the Wiki for others. BTW, I purchased the HT488 because I was told it's a direct replacement for the Supra 3000 which is no longer directly available to end users per Cisco. If it's the HT488 that's a piece of junk someone please let me know so I can return it. Thanks James Ronald ___ --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 ___ --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] Need HT488 FXO example for both inbound and outbound.
I'm new to Asterisk and I'm looking for example of how to set up the FXO side of an HT488. I have the FXS side working and can place calls between it and soft phone just fine. What I was able to find the Wiki, forums google has not been useful to me. I think I'm missing something simple probably on the HT488 device. Once I have working example I'd be happy to post it on the Wiki for others. BTW, I purchased the HT488 because I was told it's a direct replacement for the Supra 3000 which is no longer directly available to end users per Cisco. If it's the HT488 that's a piece of junk someone please let me know so I can return it. Thanks James Ronald ___ --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
