similar to: PRI Logging

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

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

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

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

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 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

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

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

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 recompiled asterisk today from CVS and I've been having a number of problems, I've read the deadlock page on the wiki and some of it sounds like that, however, the latest issue we're having it that sometimes Asterisk doesn't seem to know the PRI channel has dropped, and assumes it's still busy. However, that same channel can be used to make an outgoing call?! Has

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

wondering if anybody knows this......does shady dial work only with a zap interface or can it be configured to be used with SIP or IAX. Nauman -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, July 08, 2004 5:48 PM To: asterisk-users@lists.digium.com Subject:

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

hi all, i am trying to get my TE410P (see previous posts) working in Austria (telekom Austria - i am still waiting for an answer for my questions). my /etc/zaptel.conf looks like -------------------------------- span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=at defaultzone=at -------------------------------- after

I am trying to setup Asterisk to act like a PBX connected via a PRI gateway to a voice netowrk where Asterisk is doing outbound overlap dialing for calls that terminate via that PRI. AFter researching through the archives and online dcocs, I thought I had everyting setup right, dialplan configured for '_X!' and the 'overlapdial=yes' in the chan_dahdi.conf file, but when I try and

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: