similar to: Asterisk Setup Question -- Please Help

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

Sorry for the very newbie-like question. I have the FXS part straight. The part I don't understand is the FX0 part. Will I need the FX0 card if I am connecting to a service like FWD? My goal is to get rid of my phone line all together. I am under the impression I will only need an FX0 if I'm connecting to the central office side of the phone connection or to an existing PBX. Bottom

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

> > 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

Hi List Is there a cron that I con do to replace this, as the fx0 card doesnt hang up properly phonegc:/home/samantha# asterisk -r Asterisk CVS-05/30/03-17:17:07, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= Connected to Asterisk CVS-05 currently running on

Hi: I have a machine with four interfaces connecting four different networks. I am learning to use nmap and trying to force the nmap working only one interface. As nmap man page states, I use -e option and it would not work: nmap -e fx0 -v -sP 192.168.128.0/23 Starting Nmap 3.95 ( http://www.insecure.org/nmap/ ) at 2006-02-04 14:04 CST getinterfaces: Failed to open ethernet interface (el0)

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

I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and asterisk as pbx. I need feature called as 'three way calling' or 'transfer with consultation'. Registering,calling and 'blind transfer' work fine. Is this feature provided by sip clients or by asterisk itself ? What I have to configure in ATA and what keys I have to press on my phones ? Three way calling is

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

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

Dear all, I have an FX0 card installed in * and connected to a PBX. Calling works ok ( both in bound /out bound) but after the call, I have to press the '#' key to terminate the call, otherwise the line stays busy. Anybody has a fix for that? Thank you. Anthony

Newbie to Asterisk I've been looking around for a little while, can't seem to find some sample configs for using a Cisco 7206 as a gateway. The below link is an initial plan of an Asterisk solution that may replace our Cisco Call Manager 3.1/ IPCC / IVR setup. We currently have all of the hardware below. Just take a peak and see if there is anything that is off base. I don't know

Hello folks, I'm very interested in the Digium/Asterisk combination but need some clarification. I would like to setup a SOHO for business and home use. Scenario One: I have one analog line, 4 analog telephones. Do I need a TDM400P + 4 FXS modules (Green) + X100P? Scenario Two: 2 analog lines, 1 selective ring number for fax, 8 analog phones. Is this what I need? 2 TDM400Ps and 8 FXS

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,

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) ?=>?

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

Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP

hello,all there is a AudioCodes Mediant 2000 out there. i want to realise ip to PSTN and PSTN to ip connection. after some configuration on AudioCodes Mediant 2000, PSTN to ip connecttion works. next ,i want to dial from asterisk to PSTN now. i have see the sample in the extensions.conf relevent to PSTN as follow: ; If you are freely delivering calls to the PSTN, list them here ; ;exten =>

Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (=> one single IP with different SIP ports), the last entry into my

Just a question, has anyone knows how to blank or factory reset an AudioCodes MP124-C/FSX/AC/SIP unit (it's a 24 FSX to SIP unit). I purchased them second-handed with no manuals (thank god for the internet!!) but i guess the pdf manual I have does not have the section of factory-reset. Also, any sucess stories with: AudioCodes MP124-C/FSX/AC/SIP