similar to: How to resolve CallerID from AudioCodes FXO

hi lee. I see your problem with trunk iax, probably i don't have the solution but i don't knew if you can help me to solve mine. i want to connecte two asterisk server: server A and server B. i want make possible calls betwen all asterisk users.: users in server A with sip number 022100 can phone another sip user in server B with number 037100. this is my config:

I was able to setup AudioCodes MP-114 to rote calls form FOX to Asterisk and make internal calls: Routing Tables -> Tel to IP Routing: *, *, 10.0.0.109 (my asterisk IP) But I'm not sure how to setup AuioCodes to make calls out via FXO? In extensions.conf [Globals] pstn-5665=10.0.0.157 Whenever, I try to call out I get a busy signal. -- Joseph

Is anyone out there using an "AudioCodes MP108 8-Port FXO Analog Gateway (SIP)" with asterisk to support both inbound and outbound calling? If so, I'm interested to hear how it works, and I'd love to see some example confs (both in sip.conf and on the MP108). This product has been recommended to me by a SNOM/Asterisk-friendly distributor, but I would like a second opinion

Does anyone have configs on the MP-108 FXO to asterisk setup? I'm doing my best with the limited docs, but having very little success. Thanks, -Darren

My experience with an MP-108 was similar. Incredibly complex to setup, and very little help from MFR, or even ABPTECH, the main US reseller. We just couldn't get it working properly. Ended up with a TE110P with an Adit 600 channel bank, which ROCKS. Unbelieveably easy to setup. No echo whatsoever. -Darren

I'm trying to connect this FXO GW without any success 1) I had to configure the " " to "allow any SIP to connect , so there will be a connection. afte that when I'm dialing I get a noise. I read on the internet that I have to change the impedance (?) 2) I could not find any HOWTO configure this combination - I read that it is a lot of pain to configure the Mp10x but than

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 there! Is there anyone out there using gnudialer? I tried vicidial but couldn't get it to work (does vicidial support SIP trunks anyways?). Gnudialer seems to be simpler, though their web interface needs a little work (version 2.0seems like a step in the right direction but it isn't out yet). How do you register an agent? The documentation is lacking... Jesus Mogollon

hi we need a predictive dialer which can be used with asterisk software. Is it possible? Bye Owais Bin Zuber --------------------------------- Do you Yahoo!? Yahoo! Search - Find what you’re looking for faster. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040310/a49a4b5c/attachment.htm

Hello everybody, I'm installing an Asterisk 1.2.7.1 with GNUDialer 0.98-puff18. It also has zaptel from CVS. My FXO is an X100P Clone. The agents from GNUDialer log ok, and everything is fine until the GNUDialer makes a call, as soon as it engages (the phone starts to ring) asterisk crashes with these messages: > Channel Zap/1-1 was answered. -- Executing

Hi, Any of you have any experience with SIP softphone dialler that capable of local recording? (recording to files in harddrive) So far I only know eyeBeam and Express talk. eyebeam fine but there are known error with recording. Express talk recording looks ok, but sometime it doesn't have incoming voice with *. Cheers Benni-

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) ?=>?

It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

<!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