similar to: Call stealing

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

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

Hi, 1. When an agent is active on a call, i need the ablity for a third person to join the conversation. Basically barge in by a supervisor, participate in the conversation and then leave. 2. As an extension to the above, while on call, can the agent request a conference from another agent and later hang him up. 3. Is there any way for a call to be put in the queue destined for a specific agent

Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? Thanks for your help Robb

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:

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

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

I was wondering if anyone knew how to do the following.... Call comes in, gets put into a Queue, say `Sales`. Then the queue member is presented with the option to exit the queue, yet have the phone system sit in their place for them. When the virtual member reaches the front, call back the caller and connect them to the agent. Any ideas? Did i explain that ok? :) Cheers, Ben Merrills

Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats?

I'm looking for an enhanced speed dial "dashboard" as DSS (Manager integration) for Operator console integrated in a voip phone (softphone or hardphone, opensource or commercial) to diplay the status of phones (sip, zap, iax...) connected to asterisk. I see in snom site the snom 220 with keypad 220. Can it display the status of internal and external lines (free, busy..) and

Hi all, How are things going ? Is there a way for me to individually identify each BRI channel on the Sirrix quad BRI board. The reason I ask is because our client uses the "Asterisk Flash Operator Panel" to monitor its external lines and transfer calls from the lines to the various SIP phones. The "Flash Operator Panel" requires that we set a static value for each line or

Hi All, I'd appreciate some help on how to implement "call stealing". That is, where you dial a code to redirect any call on the system to your handset. I'm getting rid of my BRI service and I'm trying to replace the functionality of my existing ISDN2e PBX (Cybergear Gold) with VOIP and Asterisk. On my ISDN PBX, the short-code *46 does this. For example, if I take a call on

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'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

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

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

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple