similar to: Mediatrix 1102 Auth

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 all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

And if you want to use it with windows telephony software, such as answering machine or modem communications software, you can probably take the drivers for the Intel MD3200 based modem, modify the .inf for the Digium vendor and device ID. I have not tried this, but since the MD3200 modem works that way in Linux, the X100P may work that way in Windows. Then you would have a $100 winmodem! Let

Hi all. On 2 october will start SMAU, here in Italy , in Milano. SMAU is the biggest IT (and computer related stuff) expo event that we have in italy. I'll be @ SMAU from 2/10 to 6/10 , in the opensource area, where my company will promote asterisk & digium hardware. If anyone will attend the expo, drop me an email off line, so will be able to meet at the expo and chat a bit ;) Matteo.

Hi. I'm not using snom phones for a while, but now I want to test again them and I'm gonna buy a snom 200 & 105 . Some times ago I had a snom 100 , and gsm wasn't working with *. How's now the situation? the snom gsm works well with * ? Thanks for any info, Matteo. -- Matteo Brancaleoni Espia System Administrator Email : mbrancaleoni@espia.it Web : http://www.espia.it

I wish 'A(x)' was available with AgentCallBackLogin!! :( -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matteo Brancaleoni Sent: Tuesday, February 03, 2004 6:48 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Playing announcement to called user prior toConfirmation show application

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

We just did an upgrade on our Asterisk to the CVS version and our Mediatrix 1102s stopped working correctly. Our Asterisk is connected to the PSTN with a PRI. Calls from the PSTN to the Mediatrix 1102 work fine. The issue is calling out to the PSTN from the 1102. Asterisk looks like it process the call just fine except there is no talk path. Get this, though: If you flash hook and then

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

Hi, I have problems with Asterisk recognizing dtmf tones sent by Mediatrix 1104 FXS. I can not enter mailbox number (voicemail) or pin code (meet-me). Asterisk shows 'username not entered' when dialing in voicemail. Both asterisk and Mediatrix have set inband dtmf. Can anyone help me out ? Best regards, Dave

Hi all, I just got a Mediatrix 1104 evaluation unit -- a 4 port fxs sip gateway. There's no printed documentation shipped with the unit, but I have a piece of software for windows that shipped with a different model (which I haven't tried configuring yet), that uses snmp to set misc variables (ip settings, sip stuff, etc.). Fairly baroque interface & pretty slim on help...

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:

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 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

Hi all, I just picked myself up a Mediatrix FXO SIP gateway to play around with and hook into Asterisk but have no documentation. Are there default passwords or IP's that I need to know if I do a factory reset? Or better still, would anyone have a User Manual they could send my way? Any help would be appreciated. TIA. Jamie

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 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:

-----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

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

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