similar to: H323 with Tenor CMS Gateway

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

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 *'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

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 ?

Hi, just wondering if there is anyone that can help me configure my quintum box to operate with asterisk. i have tried and made numerous attemtps configuring the tenor to work with asterisk@home but have been unlucky. anyone out there has a cheat sheet to configure this device. thanks.. for some reason i cannot get it to work. your help is appreciated.

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;

Dear R forum, Please refer to my query regarding "Error with function". I forgot to mention that I am using "plyr" library. Sorry for inconvenience. Regards Katherine --- On Tue, 23/4/13, Katherine Gobin <katherine_gobin@yahoo.com> wrote: From: Katherine Gobin <katherine_gobin@yahoo.com> Subject: [R] Error with function To: r-help@r-project.org Date:

Dear R forum, I have a data.frame as given below: df = data.frame(tran = c("tran1", "tran2", "tran3", "tran4"), tenor = c("2w", "1m", "7m", "3m"))  Also, I define libor_tenor_labels = as.character(c("o_n", "1w", "2w", "1m", "2m", "3m", "4m",

Dear R forum Please refer to my query captioned Error with function. I had missed in bracket ")" in the return statement and hence I was getting the error. I has struggled for more than 2 hours to find out the problem and only then has posted to the forum. I sincerely apologize to all for consuming your valuable time. Thanks for the efforts at your end. Regards Katherine --- On

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",

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:

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",

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

I am introducing the scan() function to my class. Consider the following file (Scanexamp.Rnw ) \documentclass[12pt]{article} \begin{document} <<>>= height = scan() 64 62 66 65 62 69 72 72 70 part = scan(what = character(0)) "Soprano" "Soprano" "Soprano" "Alto" "Alto" "Tenor" "Tenor" "Bass"

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 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

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

Having solved this problem, I am posting this so that the next time I search for how to do this I will find an answer... Using qqmath(..., groups=num) creates a separate qq distribution for each group (within a panel). Using the 'col' or 'pch' argument does not (usually) work because panel.qqmath sorts the data (but not 'col' or 'pch') before plotting. Sorting

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.

Hi Guys, Hope fully somebody out there will have experienced this and can shed some light on how it was overcome. Current setup includes asterisk 1.6.2.11, GNU GK and a Quintum Tenor CMS on the same lan. Earlier I was unable to make a sip call from the CMS back to a sip client registered on my asterisk box. So I moved onto passing the call from the Quintum CMS to a Quintum Tenore DX which is also