similar to: Digium X100P for $43

Since i'm sure there are others out there in the same position as me, being disappointed that the original T400P and E400P cards are no longer available from Digium, I thought I would pass on a place I found to get them. I needed the older card, since I didn't want to have to get a motherboard with 3.3v pci slots on it.

101's for 68.00 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3042066051&category=11175 102's for 79.95 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3042066389&category=11175 Just passing what I find along.... bkw

If anyone is looking, I just ran accross an ebay auction for X100P Cards at what I thought was a very reasonable price. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3046843672&category=48483&rd=1 --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software -------------- next part -------------- An HTML attachment was

Is this an acceptable machine or does it have to be "rackmount" ? http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=250111166910 - rh -- Abba Communications Spokane, WA www.abbacomm.net

I downloaded asterisk and was trying to compile fresh, It end up in error, Any help appreciated. cvs checkout asterisk cd asterisk make clean make END UP with following error, (Previously I was able to compile without any errors. After a make clean stopped compiling.) gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o -lmysqlclient -lz /usr/bin/ld: cannot find -lmysqlclient

Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its nothing special but it may be of use to the newbies.. Like me.. If you would like a copy let me know and I will send it to you.. later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

I'm having issues with echo. My setup is Polycom IP501 phones connected to an Adit 600 via T100P. 13 Lines going to the Adit. All echo so far is on the local side (Employees hears own voice, but only on some calls). Watching the channel with ztmonitor, I notice that TXGAIN is pegged out most of the time. RXGAIN is anywhere between 45 and 60%. Is it safe to assume that playing around

Hello Considering how cheap PCI modems are compared to even the cheapest PCI hardware from Digium, OpenVox, Sangoma, etc.... I was wondering why Zaptel can't be used with those to connect an Asterisk server to a POTS line for low-level use? It just seems overkill for SOHO users who only get a few calls a day. Is it because those modems are usually Windows-specific, and it would be too

Did CentOS get an alpha yet How does this one look? http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=160115818814 will it do the job or not? - rh -- Abba Communications Spokane, WA www.abbacomm.net

Has there been any luck getting the 3Com NBX series phones to work with Asterisk? Thanks! -Andrew

Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

I have a problem with my asterisk@home installation (configured with AMP) My question is this, can you have different ports for different contexts within IAX? [Faktortel] port = 5036 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls allow=all ; Allow all codecs register => XXXXX:XXXXX@iax.faktotel.com/EXTEN

I live in New Brunswick Canada. The phone company is Aliant. When you set up business service here, you can go with either analog or digital lines. This isn't a T1 or ISDN. They are talking individual lines direct to handsets that they provide. They offer the digital option with even very small ( 2 - 4) number of lines. What technology could this be? Is there any way to connect such a

I dont recall the SPA-941 playing a stutter tone in the previous firmware but it is driving me nuts, anyone know where to turn it off? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com

Hi, I found E400P quad PRI card quite cheap (749USD): http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754 in comparison to te410p (approx 1500 USD ) http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P Now newer generation with HW echo canceling emerged (te411p). I'm not sure in what things those two cards

*This message was transferred with a trial version of CommuniGate(tm) Pro* In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that support user owned hardware/software. I need a 416/647 area code number. In looking at FXO cards

My setup is as follows: Handset -> Sipura SPA 2000 -> Asterisk -> VoicePulse and Handset -> Sipura SPA 2000 -> Asterisk -> Digium X100P -> POTS I notice when making VoicePulse calls (but *not* POTS calls through the X100P) that there is significant "popping" and "clicking" on the line. This isn't enough to interfere seriously with the call, and

Hello, I have been trying to get my coders to work without a conversion. I have read all the available asterisk documentation and support groups without any luck. Here is my issue. (Please feel free to ask questions if you do not understand what I am talking about.) I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if sip-server request g711) I have 2 SIP-services to

hi, i got he following error while trying to install digium cards in red hat linux 7.3. please help. [root@localhost root]# modprobe zaptel lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol cpu_raise_softirq_Rd01f3ee8 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_output_wakeup_Rcb5deb89 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol

I configured agi-test.agi on extension 111 when i dial into asterisk extension 111 using a IAX softphone and hangup while the AGI is playing asterisk crashes. Does anyone have any idea why this happens. -- regards Vikram (http://www.vicramresearch.com)