similar to: TrixBox and MP104 FXO (AudioCodes GW)

Greetings; We are trying to get Asterisk up and happy at our site-we tried VOIP using Sphere about a year ago, spent a *boodle* on expensive hardware and services from a local "expert", but it never was happy. I'm brand-spanking new at VOIP, and I've learned a *ton* getting Asterisk breathing in the last couple of days. I have three Polycom Soundpoint IP 500 SIP phones, which

Hi, This is my first experience with a mediant 1000 and an Asterisk Trixbox. the mediant has 12 FXOs and 12 FXSs, and I want to use it them all. I will have extensions connected to the FXS ports, and lines to the FXO. Can anyone guide me, please? regards, -- Guillermo Garron "Linux IS user friendly... It's just selective about who its friends are." (Using Ubuntu, Debian,

I have an MP114 2fxs,2fxo which I would like to use with Trixbox, does anyone have a setup file they can share to help me work this out. Instructions or a link I can follow - thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090726/643e5b96/attachment.htm

Hi all, I want to link three incoming Bell Canada centrex pstn lines (which currently go to an old norstar pbx) into asterisk. Can anyone suggest the most "painless" (i.e., "just works") way to do this? Has anyone used the D-link DVG-3004S four-port FXO-to-sip adapter, or the twice-as-costly Audiocode MP104-FXO-C3S? I know the Digium TDM400P or Sangoma A200 are options

Hi list, I'm trying to get ready the MP-104 FXO to use qith my box, but when I send calls I hear only dial tone and after a few seconds I get busy signal. I very appreciate your advices. Command line results and SIPconfigurations follows: *CLI>* -- Executing [7991696900 at total:1] Playback("SIP/101-09dd8918", "beep") in new stack -- <SIP/101-09dd8918>

hi, i've successfully configured 2 audiocodes mp104 fxs and fxo configured using SIP with asterisk currently i want to add another 2 boxes of audiocodes mp200 to add up a total of 4 trunks how can i combine this mgcp to run together with asterisk SIP? Thanks in advanced Steven

guys, what firmware version of audiocodes mp104 fxs are you using with asterisk? i'm having sip stack problems. ~kelvin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030916/50d5f5de/attachment.htm

> > I am looking for "how to" information / references on use of either Audiocodes MP104 or 108, or Vega 50 Gateways for interconecting Asterisk to the PSTN via FX0 interfaces. > > Any info of references / personnal experiences would be appreciated > > Stratus? THE WORLD'S MOST RELIABLE SERVERS * > Richard C. Sparacino > Telecom Technology Manager

We're trying to use some AudioCodes MP104 FXO units as gateways to Asterisk but cannot get them to reliably detect DTMF. Some landline calls get most digits but some are duplicated. Some cell phone calls get 0% DTMF recognition. Anyone with experience with these units have any suggestions? ABP Technical Support has been unable to diagnose the problem and is now sending random guesses and

I have an audiocodes mp-124C/FXS gateway that was a leftover from a failed voip experiment and I have no documentation at all for it. Anyone have a manual in pdf form they could share? -- Chad Whitten Network Administrator neXband Communications cwhitten@nexband.com 601-944-4801 Phone 601-944-4803 Fax

Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061022/6ca85b8c/attachment.htm

I have some Audiocodes units which appear to be running Linux, according to the unit's own "System Log" kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 However my contact at Audiocodes claims otherwise On 12/4/06, Yaniv Nizan <Yaniv.Nizan@audiocodes.com> wrote: > > > > I doubt that we are running Linux on the MP-202. Perhaps there is a

Is anyone on the list using an Audiocodes gateway with asterisk and SIP? I'm looking at that platform, but I have a couple of issues: 1) Echo cancellation. The echo that I'm hearing with an X100P is unacceptable. Does the Audiocodes do better? 2) Line signalling. I'm using Kewlstart with the X100P, but it looks like the audiocodes uses loopstart only. How does this work with

Is anyone on this list using and audiocodes FXO gateway? I have Asterisk(1.07 on OS X) setup and working fine, including SIP phones and IAX2 phones - I can make outbound calls just fine and receive inbound calls just fine. However, I can't seem to find the right series of DTMF settings on the AudioCodes to allow DTMF tones to be sent after an outbound call is connected(phone banking,

Hello all, I'm working on a little project right now and have ran into a snag. Was hoping someone would be kind enough to give me a few pointers to help me get past the current issue... I have an AudioCodes MediaPack MP-114 (2FXS and 2FXO... SIP firmware...) that I'm trying to get to play nice with Asterisk 1.4. I've got it to the point where the AudioCodes box picks up

Hello, I'm helping a colleague (*) which has the following setup: ITSP --- <T.38 capable PJSIP trunk> --- Asterisk 13 --- <PJSIP>-- Audiocodes MP-112 --- <FXO/FXS> --- Fax machine My issue is the following : Audiocodes gateway reject INVITEs with 488 Not Acceptable Here It seems this gateway requires t38 settings to be present in SDP body in the very first INVITE. My

Does anyone have any information for me? Welinghton. Citando Welinghton Magno Guimaraes <welinghton.guimaraes at ufvjm.edu.br>: > Hello! > ? > I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) > to make external links. Does anyone have any manual or instructions on > how to proceed? > ? > Asterisk ?=>? Mediant 1000 (AudioCodes) ?=>?

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

I've MP-114 that is working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension. Anything longer than 3-digits is cut off, example I dial extension 1000: [Apr 29 10:03:30] NOTICE[3817][C-000000e9]: chan_sip.c:25902 handle_request_invite: Call from '54' (10.0.0.115:5060) to extension

I have an Audiocodes MP-118FXO in production. When an outbound call is made and the remote party hangs up, the Audiocodes hangs up the call immediately. But if an incoming call is received and the remote party hangs up, the Audiocodes does not hang up immediately. I have tinkered with Current Disconnect and Polarity Reversal settings, to no avail. Anyone experienced this issue with Audiocodes or