similar to: Echo: X100P vs. Cisco FXO cards

I know this is probably way out there but............ Would it be possible to set up a (Asterisk based) conference call (per se) with the presentations at the upcoming Astricon conference via IAXtel (or something similar) so that people who are not able to attend could join a Meetme conference (listen only) and listen to the content. There maybe bandwidth issues but this would certainly be an

The subject says it all. A couple of my sons have very annoying friends that tend to call ALOT. I usually don't like to answer the phone but these kids keep calling back with in 2 minutes of calling. I'm sure someone else has this problem and maybe using * to do a callerID match and block? Even add logic that if they called so many times in an hour? Or in my case, make it a

I still have echo during the first 15 seconds (or so) of each call, and the exact same thing is happening with either of two X100P in the same 2.2ghz system. asterisk# cat /proc/interrupts CPU0 0: 198457690 XT-PIC timer 1: 60 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC usb-uhci 8: 1

Hi, I need to know from your experience which is the minimum hardware configuration (proc./memory) to run * with two X100P cards and around 10 internal IP phones (from which 4 are hardware ones and use G.711, the rest of them with GSM), with a couple of IAX connections with other Asterisk PBX'es (GSM too). Have someone tried to build a graph about the dependencies between hardware

Hello, The harddisk of my etch/bristuffed asterisk1.2 box finally died. I moved the cheap (1397:2bd0) HFC-S card to a squeeze host (i686) and built dahdi modules 2.3.0.1 using m-a. After zaptel->dahdi and asterisk 1.2->1.6 config adaptations, everything seems ok, except for the BRI side, unable to bring layers 1/2 up. Asterisk reports: chan_dahdi.c:12393 dahdi_pri_error: 1 Unable to

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Does anyone know if you can put two of the X100P cards in to the same machine and have access to two landlines ? I just need to know if it's worth buying two or not ! Thanks, Paul. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040410/2608a3fd/attachment.htm

Good news! FYI, worked with Mark this afternoon to test changes needed to reduce or eliminate echo involving pstn calls on the new tdm fxo card (bug #1902). Three simple source code changes (testing-only at this point) resulted in zero detectable echo on all incoming and outgoing tdm pstn calls, even in the first second or two. I've not noticed any unusual side effects at all. Since the

Hi, I'm thinking of purchasing a TDM400P card and was wondering if anyone has experienced any echo issues with phones off these cards connecting to the PSTN via the X100P cards? I have had my fingers burnt with a Voip phone > X100P. Cheers, Taff. --------------------------------- ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself -------------- next part

I've been using pxelinux for automated installs, and it works well. I'm using it on SUSE Linux 8.1, and it has been going well until recently. The TFTP, NFS, and DHCP servers are all on the same system. It uses PXELINUX 1.69 (part of SYSLINUX 1.62 I believe). It installed systems to my slimline Compaq Deskpro Pentium IIIs just fine. Recently, I ran it on a Compaq Deskpro Pentiums

Hope this list is for ices support too :) When I try and start ices with a simple config file it barfs and isn't happy [root@deskpro bin]# ./ices -c /usr/local/icecast/etc/ices.conf Cannot use config file (no XML support). Ices Exiting... But if I pass basically all the same paraemters via the command line it's fine streams PERFECTLY! [root@deskpro bin]# ./ices -P passwd -F

Hi all, I''ve just built Mandrake 10.1 on a Compaq Deskpro that I''ve built as a router/firewall and am redirecting port 80 outbound to force users through the Content Filter. I''ve run this setup on Mandrake 9.0 and 10.0 without any problems but this time the following happens. Squid is accessed through port 3128 and Dansguardian via 8080. If I set my browser on a

That was my thought too. Even if it *does* work without it, there may be a reason internally you can't see why it is required. (Ex. Putting extra stress on a component causing it to fail in 6 months) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Rymes Sent: Wednesday, August 10, 2005 3:49 PM To:

sorry to bother with this trivial issue, but i am loosing all my hair ;-) got 2 x100p's and * on a slakware box x-lite to x-lite works fine! i also have: #ztcfg -vvv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. and in extensions.conf i got: [locals] exten

We have been pretty much able to solve our echo problems, except for the primary mode in which we desire to operate our system. See system diagram at bottom. Prior to making adjustments to cancel echos (all echocancel=no): Call Type Result (Before) --------- --------------- CP <- LEC PRI * TDMoE * FXO -> AP

I'm not sure if I am wording this correctly, but I'll try. I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap analog phones plugged into the FXS ports. I am able to get * to ring those phones when a call comes in, but I cannot get the phones to dial out. I guess it's all syntax that I'm doing wrong. Does someone have a couple small snip-its to accomplish

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

Hi We are trying to get better sound quality out of the prompts on our Asterisk system. We had some new ones made by thevoice.digium.com and they are in WAV format instead of the default GSM format on the Asterisk server. The problem is, when you dial in to the server using the FXO card (X100P) you really can't tell the difference between the WAV prompt and the GSM prompt, however, if you

Hi I have the X100P card which as to sockets (LINE - for fxo line ) and PHONE (to connect a n analog line) This card is setup as fxs_ks I was getting dial tone but suddenly no Dial Tone....Help appreciated.... When I try to route the Call using - Dial Zap/1) to this FXO Line I get this error: ------------------------------------- linux*CLI> -- Executing

Is the X100P FXO PCI Card capable of detecting a fax, answering the call, and then emailing the fax content to an email address? Thanks. Paul.