similar to: mediatrix, dtmf

Hi, I can not send voicemails as an attachement. When setting the "attach=yes" option in voicemail.conf the mails get rejected from the mail server: ----- Transcript of session follows ----- 451 4.4.1 timeout writing message to higgs.elka.pw.edu.pl.: Connection timed out with higgs.elka.pw.edu.pl. 451 4.4.1 timeout writing message to elektron.elka.pw.edu.pl.: Connection timed out

Hi all, I am planning to use VoIP gateways to connect remote offices to Asterisk. Not having much experience with these and Asterisk I would appreciate any info/experience that you might share with me as to their interoperability with Asterisk. I am interested with in rather bigger gateways (order of E1's) from: AudioCodes - Mediant Mediatrix - 1531 Quintum tenor Multupath D3000 Has anyone

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'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

Hi, Since I've upgraded my * to 0.7.1 I see no new cdr's in my MySQL. Does this new version of * only work through ODBC ? Do I have connect to MySQL through ODBC now ? Regards, Dave

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 all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

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...

Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite <-------> Asterisk -------> PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data

Hello, When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway, there is problem with DTMF "out-of-band". See debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May

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

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

Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea?

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 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