similar to: Voice quality of Softphones vs. IP Phones an d Gateways.

I've tried almost any softphone available on the market with many different PC, soundcard, headphones combinations. None of them prooved production reliable in a call center environment. I've also tested many IP Phones and Gateways. Even the cheapest one supplies much better quality. Is this a fact or am I missing a point. I would certanly prefer a softphone because of cost and

Could anyone quote a price for the following project. We should be able to generate a specific (say 12Khz) signal at certain intervals (calculated using a price/rate table on a mySQL database) DURING an ongoing conversation. The conversation is to be marked (start and end) with specific signals as well. This is a requirement for special hotel applications where a device counts the signals to

I would recommend that you look at the Pika Technologies Daytona MM board. It has onboard DSP and onboard analog bridging taking up much less horsepower. Please contact me off-list if you would like more information. Bill Hunt Stroudwater Contact Point 207 347 8080 x219 877 870 1234 Toll Free www.stroudwater.com "Realize the Value of Customer Contact!"TM This e-mail is intended

What they are probably marketing is putting in their own equipment out there. I install a product that does exactly that. A paradyne jet fusion. It takes care of the part of which channels are data and which are voice. If it's anything like these, the lines will come out on pairs. You will then have to use channelbank and FXO/FXS cards to get it into your phone system. The jet Fusion

Can anybody please answer this. Both clients are behind different NAT's. One of them starts a SIP call to the other through Asterisk. Asterisk sets up the call. Issues reinvite and connects them together. After this point does the media stream flow through Asterisk or Peer to Peer? Does such a call use any system resources of Asterisk server after connection? Thank you in advance.

Hi! Is there anyone that are using a E1-channelbank and have any tips about some type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I think we're using some slightly modified version here in Sweden, but I'll check that tomorrow) and connect one port to a channelbank for 30 analogue telephones. It would also be great to get callerid on the analogue phones, so it would

Dear List, I was trying to verify that I could use extend.series in the wavelets package and kept getting an error when trying to use method="zero". I'm not seeing where my syntax has gone awry. According to the documentation, [see ?extend.series] " method: A character string indicating which extension method to use. Possible values are

Does anyone have any recommendations on how to connect 160 analogue phones to an asterisk PBX? Background information: A client wishes to replace their current PBX with a new VoIP system. Currently they have 2 PRIs. I intent to set up 2 asterisk PBXs with Debian GNU/Linux on raided drives. These drives will be mounted only read-only to recover gracefully from power-cycles. I am considering 2

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This package provides the headers and specification documents defining the X Window System Core Protocol, Version 11. It also includes a number of headers that aren't purely protocol related, but are depended upon by many other X Window System packages to provide common definitions and porting layer. No code nor headers were changed in this

I have just upgraded two of our samba boxes to 2.2.8 and ended up with partially broken winbind after the upgrade. The machines are slightly different, and so are the symptoms, so here goes: System 1: Was at 2.2.3 compiled from source Feb4/02, using options: "./configure --with-winbind --with-acl-support --with-quotas". Running on RedHat 7.2, installed from SGI's XFS installer

Fellow R users, I'm using the BCE {BCE} function to run a Bayesian sediment mixing model. The aim is to find the optimum % contribution from each of the 4 source areas that can yield the target geochemistry. I have geochemistry for 4 source areas called Rat: Rat<-read.table(text="CaO MgO Na2O Al2O3 Topsoils 2.511250 0.7445500 0.7085500 14.10375 ChannelBanks

Dear R-devel list members, we have developed a package and it works perfect in R 1.9.1 but when we prooved with R-devel we get the next output from the R CMD check: Error in eval(expr, envir, enclos) : couldn't find function "@<-" Execution halted ERROR: installing package indices failed ** Removing '/home/pgomez/genArise.Rcheck/genArise' ERROR Installation failed.

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

In the dial application when configuring the Limit parameter: L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) I want to read 'z' from database, based on the dialed number. How is this possible? Cenk. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm using *CVS Head version and read the dialplan from MySQL. I'm making A-Z termination to over 4000 different country and city codes. I have 3 different dialing rules depending on the price level of the dialed number. Should my extensions table contain 4000 lines? Is this realistic? Or is there any other (more clever) way doing this? Regards, Cenk. -------------- next part

The following works fine as a standalone ruby program: ********************************************************************************************** t = Time.now starter = Time.local(t.year,t.mon, t.day) + (24 *3600) ender = starter + 7 * 24 *3600 lat = 40.352039 lon = -74.191961 require ''soap/wsdlDriver'' params = {:maxt => false, :mint => false, :temp => true, :dew

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

I have 2 * boxes. 1 has 2 PRI's from the Telco, and a PRI to the 2nd * The other has ZAP channels to Channelbanks for endusers. If someone on the second box calls a Toll Free number (it probably doesn't matter that it is toll free) that is auto answered by an auto attendant (QVC, a Bank, the Airlines, Credit Card Companies....) then the call gets dropped with in a couple of seconds of

hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar

Hi! I have a E1 PRI connected to my TE400P card on span 1, and two channelbanks on span 3 and 4 and * 1.2. Every few hours I get this message and asterisk dies just after that: Warning: No D-channels available! Using Primary on channel 16 anyway! When this happens restarting zaptel and asterisk services, generally puts the system back online my zaptel.con reads: span=1,1,0,ccs,hdb3