similar to: Virtual queue member

Has anyone tried integrating Asterisk and Rhetorical's rVoice software? We're evaluating different approaches to system announcements via T2S. Has anyone gone down this route that could give some advice? I've installed festival and wasn't too impressed, the demo one the website seems far better quality and clarity then the defaults in the source package. However I must admit

Is there a way of getting the dialled number from an AGI? Is it passed in the initial variables, or can it be pulled out or passed across from the dial plan? Cheers, Ben Merrills Griffin Internet -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040630/71e0bcda/attachment.htm

Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone who's also interested, contact me off list. Cheers, Ben Merrills -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've been doing some work on a queue log analyser for a while now, getting the basics in place, an example of which you can find at the URL below. However, just wondering what information people think is most useful in a log analyser? At present it includes the following features: # Time periods - specify a period of days from the log which you want to generate statistics for (e.g. only the

Is there a way to log all PRI events to a logfile? Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com <http://www.griffin.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/d983141f/attachment.htm

I don't know if anyone noticed my post a few months ago on the asterisk-user mailing list, but I've been writing a queue log parser. I was wondering if anyone had any queue_logs (the bigger the better) that I would use as sample data? I would of course be willing to post the stats up for the people who send me their sample logs. Post them to me off list, and please tell me if you are

Hello, We are looking for commercial solution SS7 with Asterisk. It does not need to be "build-in" with Asterisk. Could anybody suggest something? Thank you in advance. Bart

Hi, I would like to know how suppress number for outside dialling in CDR table. For example, if I need press 9 key to make an outside call, I would like that the number in dst field in cdr table was the outside number without 9 key. It's possible? Thanks in advance, srsergio

Hi, How can I (through the manager interface) steal a call from one phone, and transfer it to another? Does asterisk provide for actions like this? It's a common action in Lucent systems it seems. Cheers, Ben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040816/7a79aa0f/attachment.htm

Is it possible to select the audio stream that's played as a user enters a meetme conference? If you could, it would be very simple to record a users name, and then play that as the greeting to other attendees as they join the conference. If not, could someone tell me how hard it would be to modify the source? I presume at the moment the file to be played it stored in a var somewhere,

I've got 4 Cisco 7960's and they're behind a firewall (sonic wall). The asterisk box is on a WAN connection on the other end of a DS3, the phones connect fine to the Asterisk server as you can see from the output of show sip peers below. tp3/tp3 <firewall-ip> D N 255.255.255.255 60665 Unmonitored tp2/tp2 <firewall-ip> D N

Is it possible to record a meetme conference? What channel would you monitor, is there a main channel that all audio goes too? If so, is it possible to use the ast_monitor (iirc) to record that channel? Cheers, Ben -------------- next part -------------- An HTML attachment was scrubbed... URL:

To those who are/were interested :-) Ok, I've setup a mailman mailing list. I don't so this often, so I've done a bad job of it, please let me know. Oh, and I know the SSL cert has expired... was self signed :-) Now I've apologised for my crapness, the URL to signup to the mailing list is: https://xdev.net/cgi-bin/mailman/listinfo/asterisk-uk Hope this is ok with

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

We'd be more than willing to test :) Our PRI provider has been trying to push us over to SS7 for ages. So we'd be very interested in assisting you. Ben -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve Underwood Sent: 12 January 2005 12:49 To: Asterisk Users Mailing List - Non-Commercial Discussion

http://www.voip-info.org gives: Warning: mysql error: No Database Selected in query: select `name` ,`value` from `tiki_preferences` in /var/www/html/tikiwiki-1.8.2/lib/tikidblib.php on line 133 Values: Array ( ) $result is false $result is empty Was going to grab a link to give to Florent regarding his CTI thread and question about how to program against the Asterisk API...

Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(silence/5) exten

hi I really want a good queue_log analyser, but I don't want to waste > EUR 1000 for something like that, so I thought starting a small project for it. I started off in php just creating a basic parser for the log, and I'll go on extending it. See http://karlsbakk.net/asterisk/scripts/queue_log_analyser-0.0.1- pre1.php.txt Does anyone want to join this effort? roy

I downloaded the latest version of asterisk (and I might add I've also tried the stable branch too) via cvs and tried to compile it. Now, the compilation does succeed, and an asterisk binary is produced, however when I try and run asterisk I get the following error. Could anyone shed any light as to why? I've attached a copy of the make log so you can see if there any problems. There does

The feedback we are getting so far has been excellent! As more is decided the list will be updated, if you'd like to be involved in helping, please join us on the IRC channel, #asterisk-uk on irc.freenode.net. If your company would like more involvement with the event, please email me directly. I would really like to hear from people/companies who would like to: - # Exhibit a product or