similar to: dial out via fxo gateway

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

>From your experience, would you recommend purchasing 8 Sipura 3000 1 port FXO gateways or 1 Audiocodes 8 port FXO gateway? The way I see it, the advantage of going to the Sipura solution is that it is more scalable (ie. I would only need maybe 5 in the beginning and then add one by one as the needs grow) and seems to be cheaper: ~$800 for 8 Sipura's versus $1300 for 1 Audiocodes. The

I'm currently looking to deploy an Asterisk server using an FXO media gateway to connect to the PSTN and was looking for any user experiences that may aid in selecting a gateway. Specifically i'm looking for a 4-port model under 500 dollars. Within this category exists: MediaTrix 1204 Grandstream GXW-4104 AudioCodes MP114 I've read over Voip-info.org regarding these products and

I bought a couple of Mediatrix 1204's a few of months back. (Perceived advantages were relatively low overall cost and size per port, and it isn't nearly as vibration sensitive as a PC would be.) Rich Adamson's review from Feb 1 is comprehensive, and the only thing I'd like to add is this: One "feature" of these units that absolutely infuriates me is its behavior for

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've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank & T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and

Does anyone have any experience with the Audiocodes MP-108 FXO gateway? I'm looking to get one for incoming PSTN lines. In particular, does it pass caller ID information to Asterisk? I currently have a Mediatrix 1204 but Caller ID does not work, even though the specs say it does. All it sends are the names of the ports set up internally on the gateway (ie. "pstnline1" etc) when

All, I am looking at a little support on this, as I haven't found it on google yet. I have had this work on Callweaver, but am moving to Asterisk for a variety of reasons. My dial plans, and everything else transferred perfectly, though I am not sure they are 'correct' for Asterisk 1.6.1, with simple things like SIP users outlined in the sip.conf file, not in the users file,

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

Hello, I finished an Asterisk installation this weekend, and I'm experiencing a problem when a user hangs up on a line after leaving a voicemail message. I found two similar issues when reading through the archives, and have not been able to resolve my issue from their answers. http://lists.digium.com/pipermail/asterisk-users/2004-April/042453.html

Hi all! Can someone help me solve this? I've tried a lot, but with no success :-( This happens when accessing samba server locally. TIA Marcus Uerlings

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

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 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

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 angel. it is about the CallerId, i have the same problem, did you resolve it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070321/5a5ff127/attachment.htm

Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 "Unsupported Media Type" back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

-------------------------------------------------------------------------------------------- Originally posted at http://forums.digium.com/viewtopic.php?t=18045 -------------------------------------------------------------------------------------------- Hi! I'm trying to configure a DISA setup (Asterisk 1.4.11). Only, executing DISA seems to prevent any DTMF detection capability when using

Various linux distros come with xen supported after a fashion with xen patches developed for one kernel hacked into the kernel they want to use (with varying degrees of success :-). We''ve been using xen for a while at work to try and provide development and testing platforms for a slew of different linux distros to check out our software under lots of different conditions. The

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>