similar to: Configuring SPA-3000 As A Trunk

I have written up an guide on how to do bulk provisioning of the Linksys phones and ATAs. http://voipspeak.net/index.php?option=com_content <http://voipspeak.net/index.php?option=com_content&task=view&id=73> &task=view&id=73 Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 -

The client's needs are the mother of invention. We have a client that currently uses a Cisco Call Manager and one of the features they love was the Locate-Me function (or follow-me, or find-me, whatever you want to call it) which basically rings their desk phone a few times then plays a short message, and then rings their other remote phones and cell phones. The customer wants this same

Has anyone actually used these USB speakerphones http://cgi.ebay.com/SKYPE-USB-Conference-Speakerphone-Headset-free-VoIP_ W0QQitemZ9717357487QQcategoryZ101246QQssPageNameZWDVWQQrdZ1QQcmdZViewIte m Seems to get a pretty good review here http://voipspeak.net/index.php?option=com_content&task=view&id=39&Itemid =27 But looking for real world feedback. Cheers,

Hi everyone, i am face with an asterisk use management interface, at the pressent, i am using AMP (asterisk Management Portal: http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 ). Does anyone know a better and more documented management interface for * ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have d/l the iso (*@home 0.9) , built the * box and followed the directions in the * handbook and http://www.geekgazette.com/index.php?option=com_content&task=view&id=2&Itemi d=26. I created extension 200 and verified that * was running fine. Loaded Xten lite, setup the proxy for local ip (10.0.0.201) per the handbook. After turning off the Norton Firewall protection, I am able to

Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal

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I have tried everything to get BV working outbound. All worked fine until the BV change last week. I called BV and they changed me to sip gen with a new password. I stripped my Asterisk server to one phone on Zap/1 until I get this working. The same BV account works fine with a SPA-3000 so I don't suspect a firewall problem. Symptoms: Asterisk registers with BV Ok Incoming calls work

I'm still not getting my outbound to work. I've seen two patches relevant to broadvoice for chan_sip.c which apparently have already been added to CVS. I'm dropping all outgoing calls after ~30 secs. Asterisk doesn't seem to know they're gone though. I called my cell w/ broadvoice and turned on sip debug AFTer the call had physically dropped: *CLI> sip show registry

The UK Linux & Open Source Awards for 2006 will be hosted on Wednesday 25 October 2006 at the Millennium Gloucester hotel, Kensington: http://www.linuxawards.co.uk/ The CentOS Project would like to thank all our CentOS users (especially the readers who picked CentOS as their favorite distribution) for getting CentOS in the top 3 (short-list) of all Linux Distros (with Ubuntu and SUSE).

Hi, You can read here ( http://voiptoday.org/index.php?option=com_content&task=view&id=413&Itemid=136) ADA features. It seems to me that the provided Download Now link relates to a version in which some features are missing. Among those are CRM and TAPI supports. Do you confirm ? Have you heard plans to sell or provide an enhanced or a newer version that would pack those 2 features

Hi All, Just wanted to announce that MyOSS Magazine - Edition 3 (We''re Edition 3!) is now officially hitting the street. This is a community driven project which aims to publish monthly. (Sorry if this is considered as spam). Securing your Network Connection using OpenSSL OpenSSL is a very important protocol in this day and age. The profilteration of the internet has made this

Hi, I am trying to create an R interface for Dynare++ [1], a k-order solver for rational expectation models. I would like some advice on how to glue the C++ code to R. In C++, it works the following way: - a DynareModelEq object is initialized with the formulas, parses them and performs k-order symbolic expansions - the user calls a member of this object multiple times with parameter

Can I emulate SNMP windows-service using wine? I explain for what - I wont install PlotBase PM (for print-scan machine Ricoh 240W) on CentOS 5. Now PlotBase PM install on Windows XP and I have problem with limiting TCP/IP connection. I don?t wont install Windows Server 2000 or Windows Server 2003 only for this purpose and have idea install PlotBase PM on Linux.:) Can I solve given problem using

Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Anyone using Simpletelecom with *? I had a nice working system with them, my credit can out so I apply another $5 to continue testing. Since then nothing has worked. I always get: -- Executing SetCallerID("SIP/line1-74ac", ""myname"|<>|a") in new stack -- Executing Dial("SIP/line1-74ac",

Greetings. I did some digging with Google, the wiki, and on the archives, but didn't find a recent conclusive answer. If this is answered in the wiki or archives somewhere, please point me to it. I'm in the process of setting up an Asterisk box for home use. I've got a X100P card on the way. I've not decided what analog adapter(s) to get yet. The only phone service to hook up

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who