similar to: Quintum configuration ASM200 Analog 2 tenor port

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with

Hi, I'm having problems to register Quintum Tenor DX 3020 on a Asterisk box with SIP. Asterisk always returns "Username/Password mismatch". I've tried all configurations that was on the Quintum's manual, but no success. I've tested the same username and password with a Linksys (PAP2-NA) with the same asterisk box, and it worked fine. Where is the problem ?

Let's try this again. I have a Quintum AX Tenor gateway sending calls to Asterisk from BT analogue lines connected to FXO. The agents hear an echo on their side but incoming callers hear the conversation fine. I can't seem to find the problem. Anyone seen this issue before? <p style="margin: 0; padding: 0; border-collapse: collapse; font-family: Tahoma, Arial, Sans-Serif;

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio? i have a scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for intranet no PSTN at all just only IPphones connected through ehternet port and analog phones connected on FXS port.Is

Please has anyone on this list had experience with getting Quintum equipment to talk to Asterisk? Specifically a D3000 in my case. It is refusing to register and I'm out of ideas. Any help appreciated. Neil

Hello, Maybe some of you tried the SIP support recently introduced by Quintum in their AS devices. I have one Asterisk machine connected to PSTN via E1. It works properly. On the other side I got an ADSL line, with NAT and few devices behind it, like computer with X-Lite client installed or mentioned Quintum device. It works great - calls initiated from there are OK, as well as PSTN originated

Hello all, I post this issue thinking too that could help other people on an asterisk deployment over distributed offices considering both quality, prices, devices and so. Well, i am working on a deployment of a telephony system based in asterisk. My company have a central office with seven remote offices connected all through a VPN. To reduce and evaluate costs i consider solutions like:

I just upgrade quintum A800 with new SIP firmware ---------- Product Name: Tenor Analog A800 Multipath Switch - 8 ports (Rev. B) Gatekeeper Status: Mini GK Calls Allowed: 8 Feature Bit Status: -PS/+RB/-ER Languages allowed: 1 Serial Number: A002-00308F Ethernet Address: 00-30-E1-00-30-8F IP Address: 10.101.0.10 Subnet Mask: 255.255.255.0 Default Gateway: 10.101.0.1 System Software Version:

Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, _________________________________________________________________ Expand your wine savvy — and get some great new recipes — at MSN Wine.

Hi all, I was wonder what experiences there are out there with using RAID-X for performance increases. I do use RAID-1 (2 disks) but am interested in attemtps to gain higher R/W performance. Do the RAID-5's etc give noticeable performace increases? A significant help for me was using ccache for compiling programs. That was a real performance increase. Thanks for any suggestions/opinions.

Hi all, Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones. What would you recommend to use as the 8 FXS switch? I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it is the best solution. Does anyone have a better solution to build this system? If an analog

Hello! I written several implementations in R of R?my's algorithm for generating random ordered strictly binary trees on 2n+1 vertices. One implementation involves manufacturing character strings like: "X <- list(0,list(0,0))" for the case n=2. If I perform the following two steps: cmd <- "X <- list(0,list(0,0))" eval(parse(text=cmd)) then X becomes a true

Dear R forum I have data.frame as df = data.frame(rate_name = c("USD_1w", "USD_1w", "USD_1w", "USD_1w", "USD_1m", "USD_1m", "USD_1m", "USD_1m", "USD_2m", "USD_2m", "USD_2m", "USD_2m",  "GBP_1w", "GBP_1w", "GBP_1w", "GBP_1w",

Hi all, I am planning to use VoIP gateways to connect remote offices to Asterisk. Not having much experience with these and Asterisk I would appreciate any info/experience that you might share with me as to their interoperability with Asterisk. I am interested with in rather bigger gateways (order of E1's) from: AudioCodes - Mediant Mediatrix - 1531 Quintum tenor Multupath D3000 Has anyone

hi, maybe someone out there already has some experience and can help me. I have just ordered an E100P card from Digium, I already have a basic asterisk setup up & running. My application is the following : I want to accept incoming calls from the PSTN to Asterisk, and without asking anything of the client just pass them immediately to a call gateway in USA, actually we are planning to use

Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Dear R forum I have a data.frame df = data.frame(currency_type = c("EURO_o_n", "EURO_o_n", "EURO_1w", "EURO_1w", "USD_o_n", "USD_o_n", "USD_1w", "USD_1w"), rates = c(0.47, 0.475, 0.461, 0.464, 1.21, 1.19, 1.41, 1.43))   currency_type     rates 1      EURO_o_n   0.470 2      EURO_o_n   0.475 3       EURO_1w   0.461

Dear R forum, (Pl note this is not a finance problem) I have two data.frames as currency_df = data.frame(current_date = c("3/4/2013", "3/4/2013", "3/4/2013", "3/4/2013"), issue_date = c("27/11/2012", "9/12/2012", "14/01/2013", "28/02/2013"), maturity_date = c("27/04/2013", "3/5/2013",