search for: 3000's

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pic...

The party's over folks, the new official cisco/linksys/sipura policy is to no longer sell SPA-3000's to end users. Buy them while you still can :-( -Dan

Does any body know if it is possible to group SIP channels just like it is possible with Zap channels? I have a group of FXO gateways (Sipura 3000's) and I would like to treat them as a group the same as I would Zap channels. Does anyone know if this is this possible?

Hi all, We have several sipura 3000's working well for outbound calls, however the issue we have is that when calls are sent to the Sipura with Dial(SIP/${EXTEN:0}@sipura1) the Sipura does a SIP answer immediately and then proceeds with the call "in band" therefore sending dialing sounds back to the caller. Other SIP ga...

Hi, I'm new to nut. I have some 100 APC Smart-UPS 2200 & 3000's and I'm trying to do some calibration test on them. I've installed nut, with driver = snmp-ups mibs = apcc in ups.conf. I can read the variables using upsc. However it seems, I can't actually execute commands on the UPSses this way. ( upscmd some-ups calibrate.start ) Do I m...

I have installed two Sipura 3000's on my office pbx as a test, they work well an have some great features including fax detect, but I was hoping to allow incoming faxes on the FXO port to be detected and passed through to the FXS port. Am I mistaken or does this work, and how? Chris Mason Anguilla

...whisk me off to....but I digress...you will have serious interrupt issues trying to get 4 cardss working in one system. I am surprised that you would fork for a PRI card but use cheap winmodems for analog lines. You will have much better luck tossing the "x100p" cards and using either SPA-3000's, a TDM400, or a Mediatrix 1204. Kerry Garrison Publisher - http://VOIPSpeak.net (949) 502-7819 x200 - kerryg@techdatapros.com http://www.techdatapros.com > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.co...

After trying everything under the sun to get rid of echo on my X100P, I'm curious if anyone managed to solve the echo issues by switching to a SPA-3000? As well, if you have multiple SPA-3000's, can you create dial-out groups similar to the Dial(ZAP/g1) functionality? Thanks. -- Mike Benoit <ipso@snappymail.ca>

I have a potential customer who has an existing PBX with analogue FXS ports connected to phones. He wants to allow a single remote worker to be connected to one of the analogue extension ports using VOIP. I know I could do it using Asterisk with an X100P card, but that seems a bit overkill. Does anyone know of an analogue->VOIP adapter that has an FXO port in it instead of just an FXS port?

Does any body know if it is possible to group SIP channels just like it is possible with Zap channels? I have a group of FXO gateways (Sipura 3000's if you must know) and I would like to treat them as a group the same as I would Zap channels. IS this possible?

Does any body know if it is possible to group SIP channels just like it is possible with Zap channels? I have a group of FXO gateways (Sipura 3000's if you must know) and I would like to treat them as a group the same as I would Zap channels. IS this possible?

Hello, I've been experimenting with Asterisk 1.0.6 and a Sipura SPA-3000, and I've run into a couple of questions I haven't yet found clear answers to: It appears that the SPA-3000 has no call progress on it's FXO interface? Asterisk considers a dial() as answered when the SPA-3000 has dialed the number on the PSTN line, not when someone has answere...

...it locks one down. The clone X100P cards often get a shared IRQ, even if vacant ones are available. I've found both the Digium card and the clones very frustrating with this system. Thus the reason Digium does not recommend it. I've gone to a mostly SIP configuration using Sipura SPA 3000's. For analog stations, I got an Efficient Networks SB510 (H323 protocol) on EBay but it doesn't work very well. Caller ID is incomplete between it and the Asterisk for inbound and outbound calls. (No caller-id time transmitted to phone, only number sent and a default "SB510&quot...

Hi, One of my clients, wants to use * box to run weekly meetings between remote locations over the internet. What would be the best configuration for this? We are talking about two conference rooms. I am referring to the actual hardware/software and bandwidth requirements for this to work well. I have run two software video phones and I had marginal results with it when displayed on large LCDs,

I had CC readers going over the internet (with pings over 80ms) connected to Linksys PAP2. It was only successful once every 3 attempts. I had 100% reliability when it was connected on LAN. Timing is an issue, if you doing everything on LAN it should not be a problem. Just make sure you use G.711 protocol. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com

Hello list, I have several SPA-2000's and 3000's scattered about the Internet (all behind NATs). Because I do not qualify as an ITSP, Sipura will not license their "Sipura Profile Compiler" so that I can have the units remote upgrade, remote re-configure, etc (via TFTP or HTTP). This is extremely annoying. Right now if I h...

I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [root@asterix root]# modprobe zaptel [root@asterix root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel

Hi... I have the next problem.. I have a FXO card with i can make calls but i cant recive calls. At the consol, i get the next error: -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel

Hello. I am going to be putting together my first * system using FXO/FXS interfaces. All the systems I have set up thus far have been pure VoIP setups. The system I need to set up should have 3 FXO interfaces and 1 FXS interface, as well as several SIP phones. I have noticed people complaining about Digium's TDM cards - are these isolated incidents or are these cards unreliable? I intend to

...m, and they asked me an Ethereal view of > network traffic. > > Thanks a lot for your support. > > RODOLFO > It truely is strange because I have G729 running on at least 4 asterisk systems used with Cisco 7960's, Polycom IP 300's, IP 600's, Sipura 2000's and 3000's. I have never had problems. -- Kristian Kielhofner ------------------------------ Message: 7 Date: Wed, 22 Dec 2004 22:20:42 -0500 From: Karl Brose <khb@brose.com> Subject: Re: [Asterisk-Users] hint extension and Snom phones - CVS or stable? To: Asterisk Users Mailing List - Non-C...