similar to: Recent debian packages?

Our windows users are looking for a simple application to permit dialling and transfer from Windows desktop (or web page). I've looked at everything mentioned in the WIKI, and most are either not appropriate, or are not maintained any longer. I've used Flash Operator Panel, and quite like it, but I don't believe there is a way to have a per-user view (so people can only manage

Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the " == Parsing '/etc/asterisk/manager.conf': Found". (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) There is a very old feature request about this

Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel

Hi all, I want to be able to see the status of my Agents on a web interface. I have no idea how to do so. I have found a few sample script to communicate with queues manager to view queues.But I couldn't find any on viewing the agent status. Could anybody give me a clue? Regards, Pim

Hello all, A client of us, needs a queue monitoring system. In realtime he needs to now the PRI status, the agents logged in and logged out, the number of received calls by agent, ....,etc. I am not a call center specialist and i want to find a call center software to offer to my client that fits his needs. I need a monitoring solution for incomming and outgoing calls and a queue management

i've seen that opening a socket on the asterisk server i can originate a call from one extension to another in a specific context. Is it possible to transfer an existing call from the extension ... SIP/xxx to another extension in a specific context? thanks

Dear all, I've noticed that when I have a phone registered in Asterisk, and then I register another phone with the same user, the "sip show peers" in the CLI shows that Asterisk replaced the IP of the first phone by the IP of the last one registered for that user. Consequently, if someone calls that user, only the last phone rings!! How may I configure Asterisk to be able to

These are common functions. Why "Asterisk Manager" doesn't provide commands to hold and retrive an active channel? If it must be implemented by AGI, could anyone give a direction or steps? Thanks in advance, James -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi - Am I allowed to have multiple managers logged in with the same manager username at the same time? I'm referring to the id names in manager.conf. I expect so, but just want to check to help in troubleshooting a problem. thanks -todd-

Hi, I'm wondering whether or not I should go for ODBC or IMAP voicemail storage. Before diving into details, I would be very pleased to get input form others. 1. With IMAP, is it necessary to save a copy of voicemails in /var/log files so that a user can still listen to his (or her) own voicemails with his own hardphone ? 2. How then, can you make sure to skip non-voice mails stored in the

Hello, I look at the mobicents project. Somebody has experience within both projects ? Regards Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos recherches et suivez l'actualit? en temps r?el.

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI> Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas?

Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi List, Do you know if there are any plans to improve i18n for Asterisk? The current i18n way of doing it with asterisk is very limited and most of the time does not work. For example, take voicemail: "message" "received" "at" "seven" "30" "am" might sound good in English. But: "message" "recu" "a"

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

Hi List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi->exec ("DIAL $dialstring"); my $answeredtime = $agi->get_variable ("ANSWEREDTIME"); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? I'm using asterisk 1.2.7.1 and the

Hi List, I know that the zaptel modules have echo cancellation, but is this possible to do this on VoIP <-> VoIP traffic as well? I'm toying with a SIP gateway which has apparently a terrible call quality and would like to know if there is any way asterisk can help with this. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP

With Asterisk-Java the proposed solution to connect to multiple Asterisk servers is to create multiple AsteriskManagerConnection obeject. Each ManagerConnection handles its own thread so there is no need for custom thread handing code. All you have to do is to make sure is the EventListener objects you pass to these connections synchronize access to shared data (if there are such accesses). I

Hi List, Being very new to Xen I have a few generic questions for the list, I hope to grab some useful advice and pointers to documentation. I am evaluating Xen to consolidate a few existing servers into one appliance (mainly in order to reduce power consumption, heat, and hardware failure risks). I plan to have a SER router, an Asterisk LCR router, a voicemail server, a calling card server