similar to: Asterisk success stories in small-mediumoffice environments?

I am the network administrator at a small (20-30 employee) financial company. We are in the process of moving offices and will be obtaining a VoIP phone system when we do. Right now, it's down to the 3com nbx100 series and *. Having lurked on *-user for a few weeks and having seen the nifty features of asterisk, I'm convinced. The price difference has pretty much sold my superiors.

On your web you have a link http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard To Setup Calling with Diamondcard.us and I signed up and paid the money according to Stephen Karrington it was all automated... And it was automated to take money but when you look for service hookups or information you don't get it. I have tried now for last little while to contact them for support

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

> The documentation mentions that the Digium channels can be split into some > voice channels and the remainder of the channels used for routing IP > traffic. > > Does any one have this in use in conjunction with Asterisk? Does it work > well? Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where

What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and France) ? For example: AVM C2 or AVM C4 or eicon Diva server 4 BRI? Any others? Which driver is appropriate? Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -------------- next part

Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-us&page=byoa/plus&ST=1

i'm looking at potential asterisk service providers and came across oneunifed.net i googled for opinions and feedback, but haven't come across anything yet. is anyone using them or does anyone have feedback on their asterisk support and expertise? tia, george

Hi, Does anyone know of Canadian VoIP termination providers? I have Canadian customers and would like to provide Canadian dial in and dial out (canadian callerid). Thanks!

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

HI all: Iam planning to use my asterisk box as callcenter?,any one can advice me with the best callcenter open source program based on asterisk . ? Any help will be apreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100307/116f1b75/attachment.htm

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

I have a few 7960 Skinny phones. I've edited the skinny.conf file, but I'm a little unsure as to how get the phone to figure out which ip address it should register with when it boots. How do I do that? I already have a tftp server for my SIP based phones. Do I need a tftp server for skinny configs at all? And if so, can it be the same tftp server as the SIP ones use (I'm not sure

I assume there are several people on this list that have Cisco Call Manager implementations under their belt.... We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two

Guys. How do those callcenter apps work with Asterisk where a call comes in and * send a URL and some screen popup up based on callerid or something or username or id and shows all the customers info? Anybody done that? What do you need to do that? If you are using ATAs or IP Phones, how do those integrate with the PC so the screen would popup?

Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. We have a flash frontend thats tied to our backend mysql DB. We use it for loading web site traffic data, email opens, click-throughs, bouncebacks, stats, etc. It could also be used with

any testimonials for some simple wiki software to run on centos 5.4 on an intranet? all i'm after is something uncomplicated that (ideally) yum installs, and that others can start using to start sharing useful info, nothing more. thoughts? rday -- ======================================================================== Robert P. J. Day Waterloo, Ontario,

I'm running: Asterisk CVS-HEAD-07/06/04-17:49:49 built by root@gf-002-pbx-001 on a i686 running Linux I've tried placing files (both ending in .call and not) in the correct format in /var/spool/asterisk/outgoing.... I get _nothing_. No log messages, nothing on the console, zip. Permissions seem to be correct on both the files and the directory, as well (* is running as root, for right

Hi All Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e8023f35/smime.bin

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for