similar to: Outgoing-call and enter user in Conference

Hi all, Just wondering if someone have already done something like that : SIP Client_A ---> 1)call ---> ASTERISK ---> 2)outgoingcall-PSTN-->Client_B | | 3) Enter conference | MeetMe <----------------------------' with user A Make 2 user in conference (point 1 and 2), it's definitely easy, but call

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

Hello, I have finished my basic polishing of the Asterisk GUI client I have been writing in Perl/TK and have released a first beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ I am still working on a user manual for the application, but the code works and we have been using the same basic client for the last month here at my company and it is working just fine. I'm

Hi all, i have a setup with chatrooms, several MeetMe conferences wich users can change inbetween. 10 users maximum in each room. It seems like when i have more than 40-45 users on the system at the same time asterisk drops abt 20 and continnues buisness as usual. Is there anyone else who have run inte this problem? Any solutions? It would me neat to hear about peoples experiences with MeetMe,

I was wondering if someone can give me some assistance in approaching this issue: First, the background: I have an app that i originally did in PHP that I''m redoing in Rails. It can be used by multiple clients that have no relation to each other. The client information is in a database and is determined by the hostname. Clients then have several users related by client_id. The

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Dear All, Here the version 2.2 a new version of your dear CallingCard Software !!! http://www.areski.net/areskicc-doc-v2/ Many new features have been added and several enhancements made! Newest features : - A new re-build rate-engine - LCR & LCD management (OOOOHHH YESSSSS) - Billing Increment - Progressive Rate - Scheduled Rates (days of the weeks) - Expiration rates - Buy rates

I am interested in the wellgate 38XX FXO and FXS gateways (and other similiar units). My question is can outgoing call files use these devices??? Can I fashion an outgoing call file with a channel like: SIP/WellGate-1/5551212 (for the first port) or SIP/WellGate-2/5551223 (for the second port) TO these devices behave this way? Of course incoming calls I dont see as a problem. It's asking

Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as

Hello Asteriskos, Screenshot: http://www.areski.net/asterisk-meetme/about.php The goals of this application is to control your audience/users in the conference room. That will allow you to have a visual presentation and to control the conferences over the net. A lot of changes has be made to app_meetme to keep some conferences informations into a DB and to check through if some properties has

Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd

>> It is better to stay with Postgres. If you don't want to loose your >> business stay away from MySQL. >> If you are from Toronto ( I suppose you are ), you can check my posts to >> TLUG (Toronto Linux User Group) >> regarding MySQL and Postgres. I would say Postgres is a Open Source >> Oracle. It's very stable, very scalable >> and it's

Hey all, having a terrible time with asterisk-stat -- it runs, server is fine, but some of the pages don't display properly/at all --- I think this is a code problem with them, but not sure. I thought everyone loved the asterisk-stat package? See below problems. Any ideas? Areski hasn't replied to me since.... -- Chris ----- Original Message ----- From: "Chris Earle

Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

Hi, I'm trying to install Areskicc but with no much sucess. After installing the UI I login as root/mypass, but in the left menu I only have 3 options: Main Template, disconnect and logout. When I click on Mail Template> Show mail template it shows an error page. First, is there a way to increase the debugging level so I can know what's going on behind the scenes? Second, I've

Note: forwarded message attached. __________________________________ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html -------------- next part -------------- An embedded message was scrubbed... From: Khubeka JM <jmkhubeka@yahoo.com> Subject: JE TROUVE QUE VOUS N'ETES PAS HONETE! Date: Sun, 26 Jun 2005

Folks! Let me clarify something to the Asterisk community about the CDR tool. 1) This is *not* my code to start with. I picked the original code from this forum here... http://www.voip-info.org/tiki-index.php?page=Asterisk%20CDR%20Areski%20GUI 2) The original code was not working (for most part, as the MySql portion has bugs) and I fixed this and added a few bells and whistles. 3) The

Hi all, I'm looking for a reporting solution for Asterisk CDRs. I have a small Asterisk server that will eventually have 4 - 6 trunks. the system is up and the CDRs are being written to a MySQL DB. I tried installing Areski, but had no success .. I assume it's no longer supported... the last update was in March 2005 according to this page..

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: