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I caught a thread the other day concerning Astricon and users embedding Asterisk on a Linksys or Netgear broadband router. I lost track of the email thread, if anyone is presently working with this scenario please shoot me an email. Thanks Cory Andrews ++++++++++++++++++ VoIPSupply.com PBXSelect.com ++++++++++++++++++ 454 Sonwil Drive Buffalo, NY 14225 voice direct - 716.250.3402 fax -

Just got a call from a company in Warren, MI . They recently had an Asterisk system put in by a vendor, and are having issues which need analysis and correction. They have a tremendous sense of urgency. They have about (40) users, and need DID's assigned to extensions and are having some echo issues at the site. If anyone is in the Warren, MI area, and is interested in some cavalry work,

Have a customer running a 3rd party PBX implementation based on Asterisk, not utilizing SIP inbound and outbound calls I believe are coming through a Digium TDM2402B. They are utilizing Polycom phones. They are experiencing frequent static on the line, and overall insufficient volume on conversations. They are in a bit of a rural area, I was curious if anyone thinks it could be an issue with

Does anyone know if the Sangoma A101 supports DS4 and/or RBS ! Shame on me for not knowing the answer, but I don't, and I can't find any mention of it on Sangoma's website. Cory Andrews Purchasing Manager ++++++++++++++++++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 ++++++++++++++++++++ direct - 716.250.3402 mobile - 716.907.4054 email -

I'm hosting some folks today from Sangoma for a sales training. If anyone on the list has any specific questions regarding current or upcoming Sangoma products, please shoot me an email in the next hour or so and I'll do my best to get you the answer. Thanks -- Cory J Andrews Partner / Purchasing +++++++++++++++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo,

I concur with Conrad. Cisco phones were retrofitted for SIP, whereas Snom phone are built around, and expressly for, the SIP standard. To be in compliance with Cisco regs, you are also supposed to have a SIP User license and a Smartnet contract for each phone if you abide by their program. I realize it is not difficult to obtain the SIP firmware, but if you were dealing with a Cisco authorized

I have a lot of folks asking me about an auto-negotiating WLAN phone supposedly being brought to market by Hop-On, which is touted to carry an MSRP of $40 Press photos (stock art) of the device shows it looks almost identical to devices from Zyxel and UTStarCom. I am trying to explain to folks there is no way in hell you are going to be able to buy these phones, hardware only, for $40

Don't want the opening line of my post to be misconstrued. "Anyone familiar with the BrookTrout TR1034 Fax Board, supports T.38 and PSTN fax in an Asterisk environment?" That's a question, not a statement. Curious if anyone has played with one of these as an extension of an Asterisk server for T.38 faxing, did not mean to imply it could, should or would play well with

Anyone have any firsthand experience implementing OpenSBC (opensourcesip.org)? Have a possible consulting gig referral. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candrews at sayersmedia.com <mailto:brett at voipsupply.com> Have I

Hi Guys I am new working with lumenvox products, and unfortunately I had not been able to install it properly, I follow the steps in lumenvox site and it looks like it works I mean: ========================================================= [root at pbx-millenium examples]# ./example 127.0.0.1 Connecting to 127.0.0.1 Interpretation 1: 8587070707 count=0, decode returns 1 Interpretation 1:

Anyone using Nokia "E" Series handsets with Asterisk? I'm trying to deploy some e71's and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won't allow me to save the profile, saying that devices in the same "realm" must have identical username and password. Anyone have a workaround for this

Contacted Scott Willard at Polycom this morning, he has since been reassigned to other duties within the organization. Mr. Willard's tone seemed optimistic, and he referrred me to Roger Austin, Regional Channel Manager for Voice. Roger's reply to my inquiry is as follows: "Cory, We appreciate your interest in Polycom VoIP Phones. Polycom deploys our VoIP phones with our VoIP

Ok, I have trixbox working how I want. How do I now (cheaply as possibly) get a phone number so people can call it from any number? I am just doing a prototype so just want it done cheaply so I can demo it to my supervisors. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! I need to: 1) call special number (or run special application) on mobile phone 2) establish connection between mobile phone and server 3) allow server to recognize spoken numbers (Polish language) and some other control words 4) let the server to say some short answers (prerecorded in mp3) according to some algorithm and recognized words 5) let the server to save little text file on its

Is anyone running Lumenvox on Gentoo? My asterisk install has been running like a champ for a few years now and I really hate the thoughts of changing distros just for Lumenvox. Here is my issue: The engine needs the libs from boost. I emerged boost and noticed that there were four libs that the engine were looking for that were not installed via portage. libboost_regex.so.2

Hi listers, I'm a 1.4 holdout who uses mostly Suse platforms. I bit the bullet and installed a Centos box to get Lumenvox up and running, but now see this "new" offering of Vestec that supports OpenSuse and Windows in addition to the Platforms supported by Lumenvox. Anybody out there working with Vestec that can give me a heads up on how it works or doesn't?

This Friday, February 15th, at 12 Noon EST, 9AM PST, 17:00 UTC, Lumenvox will be joining us on the VoIP Users Conference. This week, the last in a series about IVR, Lumenvox will be there to discuss and field your questions on their speech recognition solutions. http://www.VoipUsersConference.org - for info on the conference, how to connect, etc IRC freenode.net #voip-users-conference - to

Hi, Does anyone know what is going on with voipsupply? My sales guy hasn't been online in several days, their 800 number is fasy busy, as are their direct lines. And the canadian store website is down. What the heck is going on?

Hello Luca, Thank you for your response. I?m familiar with speech recognition and TTS, but new to MRCP. Yes, the 100k options is used for names in a directory listing. In the pre-MRCP support, Nuance ASR used API events/methods for the application to tell ASR when the prompt was playing and when it stopped. If ASR detected speech, it would signal an event so we would stop playing the prompt.

Hello all, how are you? I would like to know from someone uses or has used the engines of LumenVox for integration with the asterisk for voice recognition. Best Regards Josu?