similar to: Mediatrix 1102 issue after upgrading to CVS

Hi! Has anyone on the board successfully installed a Mediatrix 1102 or 1104 as a SIP peer on Asterisk? I'm trying to configure different user accounts on each FXS port, but I'm having authentication problems; Asterisk is saying the client is not authorized. Interestingly enough, I can dial a "9" and make a local call through the Mediatrix. Thanks! chris --------------

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

Hi all, Does anyone know of a method of sending a raw G711 stream to an address in Asterisk. For example, an application that takes a argument of a phone and a port. The reason? I have found a method to paging on Zultys ZIP2 and ZIP4x4 handsets. Basically it involves sending a stream of RTP data to port 3771 to multicast address 224.0.0.1. Would it need to involve me writing my

Anyone had experience with this phone? The interesting features that caught my eye were the RTP encryption, speaker "pager", and built in 4-port ethernet switch. Of course, RTP encryption doesn't exist in * yet, but it might be interesting (I've had people ask about it, but I don't know how serious they are about needing it.) http://www.zip4x4.com/summary_ZIP4x4.htm

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

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

Hi all. I'm evaluating a mediatrix 2fxs 1102. seems great (it has also supervised transfer, that's very needed in office environments and works well). the only I thing I cannot make work is the auth to my asterisk server. If I don't set a password into the mediatrix and *, I can call out, but still the registration goes wrong. using a password, nothing works. I've done some

Anyone in need of some of these? Garrett Smith Sales Executive garrett.smith@b2llc.com B2 Technologies 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix,

And it runs linux. http://www.zip4x4.com/ZIP4x4.htm Anyone seen one? bkw

Hello all, Does anyone has experience of deploying Asterisk based VoIP solution in a universitywide campus. We are at present investigating various Soft PBX for this purpose from different vendors Digium,Snom, Pingtel... We are looking at serving more than 5000 clients and we want to be very sure before taking any final decision. I would be glad to hear from members who are aware of

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

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 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

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 have the user manual, I'll send it to your email tonight when I'll be in my home. I have an APA III-4FXO too, until today I can't put it to work with asterisk. Kind regards, Miguel Date: Fri, 03 Sep 2004 16:07:59 +1000 From: Jamie Carl <geek@j-code.net> Subject: [Asterisk-Users] Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual? To:

Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich

Dear Group, I have been able to configure my Asterisk BOX to receive calls from Mediatrix 1204. I'm having problems sending calls out via my Mediatrix unit. The SIP Invite is sent to the Mediatrix but the Mediatrix unit sends back a Status : 480 Temporarily Unavailable. This is my configuration on Asterisk; exten => _78996.,1,Dial(SIP/${EXTEN:5}@192.168.6.52) exten =>

i'm new here and i need help on how where can i get software version 4.0.x of the mediatrix and how can i install it... mediatrix unit im using has a software version of 2.4.9.57. i would like to use H.323 not SIP... please need help asap!... hope to hear from anyone of you soon.. thanks in advance! ------ eder _______________________________ Do you Yahoo!? Win 1 of 4,000 free domain

Hi everyone, I just installed asterisk on my system with the purpose of rerouting calls on sip channels. I don't think i need any hardware for that. I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both of them and to call one from the other on the same machine, however could not. I 1-) I could connect sjphone in isolation to freeworld dialup howver i got no sounds

I recently purchased a Mediatrix 1124 from an auction of a company that went out of business. It came with nothing other than the unit itself. In digging thru the Mediatrix web site, and various google searches, it looks like it only supports SNMP setup, and only with their software (or the correct MIB). However, Mediatrix doesn't appear to let you download said software or MIB from