similar to: How different is implementing Cisco based system than Asterisk based system?

Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm

Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If

My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -------------- next part

Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations <sip:2001 at 172.26.48.208 <sip%3A2001 at 172.26.48.208>>;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

Hi Everyone, Has anyone got any opensource speech recognition software to work with Asterisk? Please only list WORKING ones. Not the "theoretically" should work ones! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100821/4d11d6c0/attachment.htm

Hi, I use asterisk with sip3000 device with "sip-aho" connected to PSTN and "sip-ahi" connected to a phone. When call arrives from PSTN, the *phone continues ringing even after caller hanged up*. The dialplan contains the following lines: [from-pstn] ... exten => 99,n,Dial(SIP/sip-ahi,30,g) exten => 99,n,Hangup() The asterisk properly detects hangup of the caller as I

Hi, Does anybody have experience working with Aculab groomer II, to convert between ISDN E1 and non-ISDN T1, or anything similar. I am looking for sample config files. We have asterisk as ISDN E1, but for testing we set it up as regular T1 if we get sample config files. Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current

Hi List, When doing 'ael reload' on two servers, which are setup with asterisk 1.4.22 and 1.4.35 respectively, I am getting multiple lines of this strange error: ERROR[15483]: ael.flex:647 ael_yylex: Unhandled char(s): On three other servers with same versions of asterisk, i.e. 1.4.22, I don't see this error. Number of lines of the error are the same as the number of lines of the

For a while now I've been looking for a good web based soft phone solution, but so far no luck. A few solutions which I've tried, both Java based and Flash based, either don't work, or had bad sound quality. I need something which I could put on my productions server for my clients. Seems like good web based solutions are all paid ones, nobody is giving it for free. Any ideas,

Hi, Why Aastra phones use more electricity, i.e. 48VDC whereas other phones use much less, e.g. Grandstream and Linksys both use only 5VDC. I first thought it was because of PoE, but the ones with 5VDC also run fine on PoE. What is the difference in power consumption then? -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello my friends, I want to make upgrades for all my software, currently i have the following versions: Asterisk 1.4.21.2 Zaptel Version: 1.4.11 WANPIPE Release: 3.4.7 libpri version: 1.4.5 I want to make upgrade for the last version of Asterisk 1.4, the last version of Zaptel (dahdi will be nice!), the newest libpri version and wanpipe What should i do? this is a production server and i

Hi All, I?m trying to do ?things? after my Dial application terminates (e.g. play IVR to called party, calling party, etc.). I?m trying to use the local channel for this purpose but so far with no success. I?m using 1.6.1.18 and this is my extensions.conf: [Internal] exten => _22,1,Dial(Local/${EXTEN}@CW/n) ; 22 is test number exten => _22,2,Noop(After Hangup) [CW] exten =>

Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring