similar to: Manager.conf '127.0.0.1 unable to authenticate'

Hello, >From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS.

Hi list, trixbox web-administration can be reached by host ip. since I am trying trixbox on the machine where I host my website as well, can I move trixbox main page to xxx.xxx.xxx.xxx/asterisk? which file I should move and should I modify the file? Thanks. Kurt _________________________________________________________________ Get live scores and news about your team: Add the Live.com

I have installed Asterisk from TRIXBOX.1.2.3 Please help me, How to I configure H.323 TRUNK between Trixbox and AvayaIPPBX. Uugan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061205/4fcfe45c/attachment.htm

Hi I want to setup an IVR on Trixbox and use it to send calls to agents, and i want to integrate this with sugar CRM that comes with tixbox. can some one please help me Adi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060620/66d8121f/attachment.htm

Hi every body, I've set up a Trixbox Server with TE110P,all things seem to work fine(Thank You Malling lists & irc & Forums), but i need your help, i ve 30 numbre from 60 to 89, i need to specify for each sip extension a Zap number for example to call the sales service the caller must call 555-4570 and automaticly the caller will be redirected to the 202 ( sales service ) so nobody

Hi list, I have a couple of questions related to asterisk billing and the generation of cdr logs. I've searched the wiki but have not found my answers, hopefully you guys can help. 1) When are asterisk CDR logs _normally_ generated? When the call arrives, when the call hangs up, or both? I have looked at the records created and it seems to only generate it at the time the call is

Hi - I've told lots of people about the Flash Operator Panel before, but I've never actually used it myself. I've got it all set up nicely, but I can't seem to authenticate to the asterisk manager (which is running on the same box). When I set the op_server.pl to give debug messages, it shows that it can reach the asterisk manager, but cannot authenticate: ** Asterisk

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

Hi. I've started working on a PHP-project that generates the configuration files i need based on what's in my MYSQL database. I can add, delete and edit users from the web. I can set up exactly the dialplan i need by arranging the users in a firms and groups if needed. I've also set up a java servlet so that i can get asterisk to reload by pushing a button from the web-interface. The

Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

There was some discussion in the past about which one is the best Content Management System that can be used in conjunction with Asterisk. Mambo was supposed to be the best out there under GPL. The guys who developed Mambo have a new product now - Joomla. I am using this and it appears to be better than Mambo in many respects. Read the gist about Joomla below. ------------- If you've read

Hi, I cannot seem to create associations like this: class Project < ActiveRecord::Base belongs_to :manager, :class_name => ''User'', :foreign_key => ''manager'' belongs_to :liaison, :class_name => ''User'', :foreign_key => ''liaison'' end p = Project.new p.manager => 1 trying to retrieve associated objects

Can anyone recommend a call accounting solution with rating for post paid billing that works well with asterisk using the account code or any other info from the CDR? I don't want the billing software to any phone calls for me, therefore any solution that modifies my extensions.conf is out, nor does it have to allow for customers the ability to log in to check their usage/balances. I have

Hello, Currenly I've ASterik@Home 1.5 running on DMZ. I can register SJphone there, good audio on 8200 (webmeet me calls) and i also can dial Zapata extensions. When I dial sip phone extensions nothing happens if the client that i'm calling is registred, if the client has voicemail it goes to voicemail. IMPORTANT: I get this error message on my Check Point Firewall: "sip

We are looking for an offline billing solution. We have a couple of particular requirements: 1) Since it's offline, we need to be able to import the CDR. 2) A way to support account credits based on referrals. Meaning, that if a member refers a new account, that member would get a free month of service, or similar type credits. 3) Generate invoices in either HTML or PDF format so they can be

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

Hello, I'm checking some Billing Software for Asterisk. In opensource I only know (the name, I haven't used) AstBill. What other software should I check with similar capabilities? Thank you! -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona

Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi List. I'm in need of something that will allow me to analyze cdr details either via .csv or mysql that will give me call durations as well as call costs. This is so that we can see in what areas/staff are costing what per month/week on outbound

Looking for a liitle help if anyone has dealt with this; The options on dial and queue of t (allow called party to transfer call) and T (allow calling aprty to transfer call) seem to work fine (as long as you do not confuse them with the same t and T that indicate timeout!). The problem I am having is the use of the # key to do so. Many times a caller will palce a call to an IVR that requires

hi all, got a small prblem here Running Samba 2.2.8 with a couple of dell laptops that are used either standalone or connected to a network. Both users are storing lots of pictures on their C drive which are visible through My Documents/My Pictures. The problem is that when they are connected to the network, all these files are stored as part of their profile on the server. When it gets to