similar to: Voice based FAQ and support system?

-----BEGIN PGP SIGNED MESSAGE----- Greetings, I've just about got Asterisk up and running and am wondering the following. Currently, I subscribe to both Vonage and Broadvoice and as such, I've got a Sipura and Cisco ATA186. Although I'm sure this is expressly prohibited somewhere in my service agreements, can I reprogram these devices to access my own asterisk server rather than

Hi, I have asterisk-0.4.0 running. When I make a call between an ATA186 and Asterisk using ulaw or alaw codec, all is fine. I installed the CVS version and tried the same thing but the voice is choppy. The installation was done on the same linux server. The stats on the ATA186 show no packet loss but a great number of "late packets". The stats when running version *-.0.4.0 do not

We're on track for a release of Asterisk 0.4.0 soon. I'd like to try to see to it that we have squared away our SIP implementation by then, and after that point, try to keep it in tip top shape. In general, I find that SIP is extremely fragile, and every time I try to fix one bug, I end up creating another somewhere. What I need are strategies for verifying that the SIP implementation

Can anyone recommend a client / phone that runs on Windows XP, with either a sound card or some other hardware? Ideally free, but does not have to be. Thanks...

Hi Guys, ActiveRecord is great for being able to drill down through many tables. I often locate the very "top" record, and have a render :action => "show" display that top record. Often I want to drill down like: @kingdom.phylums.classes.orders.each do |order| render :partial => "order/show" end But, depending on the user who is logged in, they see a

Even using the VBA back of Excel to create interfaces with R would make a lot of sense. Suppose I could have access to VBA macros that import and export data into R , it would be great. The R GUI series like Rattle come even closer to Excel...so a VBA _R_ExCel package might be useful to ordinary folks . Besides Excel costs money, so adding R functions to Open Office would help both of them ( if

Hi All... I have installed a single X100P card in my PC and am playing with Asterisk. The wire I plugged into the X100P has two POTS lines on it, wired on the RJ45 in the normal way. I am getting odd behavior. It seems when I dial out that the X100P dials both lines at the same time. I have two questions. First, I see that the X100P is only a single channel. Does this mean that I can

Hi there, I have been googling all morning, and have thus far not come up with a solution... Server: CentOS release 5.5 (Final) samba-common-3.0.33-3.28.el5 samba-3.0.33-3.28.el5 samba-client-3.0.33-3.28.el5 Client: OS X 10.6.3 (Build 10D578) All patches are applied. SMB CONF: # cat /etc/samba/smb.conf [global] workgroup = WRKGRP netbios name = Boleo server string = (%L) wins support = Yes

Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the & in the dial statement. i.e.) exten => blah,blah,Dial(SIP/GS1&SIP/GS2&SIP/GS3&SIP/ata186a&SIP/ata186b,25,t) If one of those lines is being used, then the user gets a really

Hello everybody, Which one is a better choice 1. Gateway device with FXO <-> SIP ( example Addpac http://www.addpac.com/addpac_eng2/addpac_product_view_detail.php?class_id=19&item_id=59 ) 2. Digium (Wildcard TDM400P) 3. Sangoma (A200 Analog FXO/FXS) All i need is to put asterisk in place with 4-8 incomming lines (ordinary POTS ). With IVR, Voice mail and International Call via SIP.

Hi All... I have been having a recurring problem with * for several weeks now. I am using it with two ATA186 devices configured for SIP. There is a USB zaptel device plugged in but to the PC but no phone is plugged into it. I have two X100P cards, each with a phone line. About once each day, * will just decide to stop answering calls from the phone lines or from an extension. The *

Hi I have a brand new ATA186 with the following firmware: Version: v2.15.ms ata186 (Build 020919a) I have been through the archives about how to configure it, but my colorful configuration web page does not have the same fields that people say I need to adjust. Even the examples on Cisco's web site don;t match. For example, I don't have the GtkOrProxy field, which is an important

Hi, Does Asterisk, provide the option within its voice mail system to be able to do the following: We want to have an option to be able to have two mail boxes on one ext. For example: we want to be able to have a French Mailbox and a English one... So I can say the following... You have reached ext 000 to continue this message in French press example 3 - and it would then go to my French

Hello All, I have searched for this error and found similiar errors, but everything seems to indicate that this should be fixed in the 2.5.5 version. rsync version 2.5.5 protocol version 26 I am running rsync with -axz, trying to sync up a large gz file : Source machine: -rw-r--r-- 1 root other 175214792 Dec 22 00:17

Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my "POTS backhaul." On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig

Guess who's here to do an Asterisk demo this week without the power supply for his SPA-841. I have an ATA186 with me. Both phones use a 5v supply. Does anyone know whether the supplies are interchangeable? Thanks in advance; sorry for the noise. B.

Sorry this may well be defined as Off Topic. I apologize in advance. I am interested in performing what I think would be a probabilistic sensitivity simulation. I've done some crude ones before in excel but I'm wondering if R can help me do it more effectively? I have a set of theoretical variables for simplicity lets use (what I think) is an easier example: I have a peg and a hole

On 11/27/2017 02:02 PM, m.roth at 5-cent.us wrote: > Pete Biggs wrote: >> - don't run ssh on 22, use a different port. > I consider that pointless security-through-obscurity. Security through obscurity it may be, but it isn't pointless. Tarpits are in a similar class; they don't help with security in the absolute sense, but they slow the attacker down, and that might

Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On