similar to: Interface * with ATA from ATA FXS port?

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

Just received: Cognigen is very proud to announce the official launch of Broadvox Direct, a new VOIP service. Broadvox Direct offers unlimited calling plans to anywhere in the US and Canada for a low monthly payment starting at $29.95 and basic accounts as low as $12.95. http://cognigen.net/broadvox/?mu Anyone know who's behind that? It's not BroadVoice, is it?

I'm looking for an inexpensive way to connect 20 analog phones to asterisk. I could get a bunch of Linksys or Sipura boxes but was wondering if there is a more cost effective way? I came across the Mediatrix 1104 and even the Mediatrix 1124 but that comes out to be almost $100/port. I might as well buy inexpensive IP phone. Does anyone have any suggestions? Thanks, Waldo

Guys, For what it's worth, after months of trying to troubleshoot issues with them, and after paying them around $2500 for setup and a down payment (it's unclear what of that will be refunded, if any) BroadVox -- http://www.broadvox.net/ -- decided to terminate our contract without any valid reason, and the only explanation they could cite was "it's because of the software

I'm proposing an asterisk configuration and considering the use of multiport SIP/FXS gateways (instead of T1 cards and channel banks). I'm looking for products similar in function to the Cisco ATA-186, but with more ports. I've seen the manufacturer's web pages for the Audiocodes MediaPack (http://www.audiocodes.com/) and the Mediatrix (http://www.mediatrix.com/) access devices.

After an extensive conversation with Mediatrx 's sales department , I stand corrected and so does the salesman who spoke to me. My apologies to Voip Supply. I understand now you never knew about the CD. Garrett Smith wrote: > I though I would post an update for everyone on what DOES and DOES NOT > come with every Mediatrix product. > > > > Every Mediatrix product,

What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers.

Hi All, Which is the best signalling to use when connecting an FXS inteface on a TDM400 to a standard telephone. I see that all examples use fxo_ks, but it is my understanding that kewl start is really designed for connections to the CO so that hangup etc. can be detected. So does it make any sense to configure a telephone for fxo_ks? Or should it be configured for fxo_ls? Regards Garry Taylor

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

if you have anyone questions about your service you can contact us at the support 978-418-7300 James Jones Broadvoice Technical Support ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Ben Wern Sent: Sat 8/28/2004 4:34 PM To: Asterisk Users Subject: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting Can anyone who is using Asterisk

Here is a letter I sent them for my $150 paper weight. Dear Voipsupply, As a small service provider, using you company for the first time, I'm very disappointed that you have removed the configuration CD that should have been shipped with the Mediatrix 2102 just to get a few more bucks. I have contacted mediatrix and they have informed me that the CD's is shipped in every 2102. If I

I added those lines to my configuration, and i just see with ethereal that my client dial and the 1204 led turn on and they started to interchange packets, im newbie with asterisk i have been trying another sip server with mediatrix that work so well, but i dont know how to set it up? could u send me all the configuration i need step by step? ----- Original Message ----- From: "Wojciech

I'm having a problem getting outbound trunking to work using asterisk and an external SIP FXO. 7 digit dialing produces the following output: -- Executing Dial("SIP/mitel-fe17", "SIP/5925660@mediatrix-1204") in new stack -- Called 5925660@mediatrix-1204 -- SIP/mediatrix-1204-645e answered SIP/mitel-fe17 -- Attempting native bridge of SIP/mitel-fe17 and

I am attempting to use a Mediatrix 1204 to interface to multizone paging from Asterisk. I have 4 different paging interfaces and want to connect each of those 4 to an FXO port on the Mediatrix. The desired result is to be able to issue some SIP dial string from asterisk, seize the FXO port on the Mediatrix and then have a speech path. I am able to place calls over the Mediatrix when it's

Hello guys, I need your help related to a mediatrix 1204 configuration. I read some of the messages that you posted in the asterisk users mailing list about the mediatrix 1204 and decided to contact you. I know that the community is not related to Mediatrix devices, but so far I have not found any other group that has work as much as you with them. I bought the mediatrix in Mexico and my provider

hi guys, i'm in need of several 24port or higher fxs device which supports sip, aside from mediatrix and audiocodes (cisco's vg248 doesn't support sip), do you have any idea who else manufactures such device? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030801/67eb12dd/attachment.htm

Hi, I have a client that have currently 400 analog phones (all wired w/ Cat3). I need multi-ports FXS interfaces but I only find 24 ports FXS (like Mediatrix 1124) but it's a little bit expensive to get 15-16 box (~408 FXS ports). Someone have suggestion to link all theses phones to "channel bank" ? What equipment to use ? Currently all 400 lines are terminated in M-66 blocks and

We have a Mediatrix 1102 hooked into the network. Both of the attached analog phones and all of their features work, but in the CLI we keep getting "-- Got SIP response 481 "Transaction Does Not Exist" back from XXX.XXX.XXX.XXX " (Where XXX is the IP address of the Mediatrix ) every few minutes. I have changed most of the settings in the sip.conf multiple times and have done

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm getting a strange problem with a Mediatrix 3631 Gateway connected to the PSTN via an E1 PRI link configured for Euro ISDN signaling. The Mediatrix sends incoming calls from the PSTN to an Asterisk server via SIP: this works fine. But when the caller hangs up, the Mediatrix doesn't send "Bye" to Asterisk, so the call is

Sorry, that name is a misnomer. However, I was excited to find that Ruby has a built in Observable module and I''m pretty bored, so I apologize in advance.... require ''observer'' # one who is observed class Celebrity include Observable attr_accessor :name attr_reader :is def is=(val) @is = val changed notify_observers(self) end end # one who