search for: mp104

...e 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 too, but it's physically easier for me to use an FXO > ethernet appliance (simpler wiring). I've had problems with hangup detection in some tests of different approaches... Suggestions appreciated! - Mike

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

I am trying to solve the fax problem by installing an E1 channelbank (Megaplex MP-104) It's a box that has 8 x FXS ports and a single E1 port. The plan was to use one of my 4 E1 ports to connect to the Telstra onramp and one to the MP104. I have since discovered however that the MP104 only supports CAS signalling and I am having trouble getting asterisk to work in this mode. Currently we have Onramp (Australian PRI) ---- port 1 of T410P --- asterisk --- port 2 or T410P --- MP104 In zaptel.conf I have span=1,1,0,ccs,hdb3,crc4 bcha...

...talking to the outside world. In our previous trial, we purchased an AudioCodes MP-104 MGCP gateway. I'm supposing that something needs to be set up in mgcp.conf for it, but what I've tried just hasn't worked. Here's what I have: ----- [general] port = 2427 bindaddr = 0.0.0.0 [mp104-1] host = 192.168.10.179 callgroup=1 context=default callerid="Incoming Line 1" line => aaln/1 ----- In the Asterisk CLI, I'm getting "NOTICE[12321]: chan_mgcp.c:1656 find_subchannel_and_lock: Gateway 'mp104-1' (and thus its endpoint '*') does not exist"...

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>

Hello-- In the interest of Symmetry, does anyone else in the world see any need for a device like the IAXy (or the SIP ones from other manufacturers, like the ATA186), but one that presents an FXO interface instead, so it can be connected not to phones, but the PSTN? murf

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

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

> > 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 > Tel:...

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 diag...

Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that