similar to: 7206 as SIP->PSTN Gateway?

I need to get your thoughts on something.. :) I am trying to create a system to process the CDR call logs for department accounting.. I think there are two ways of doing it.. Either I can create an AGI that will run on the "h" extension and will lookup the last entry that matches the account code of the call that just ended in the MySQL CDR and calculate the call cost immediately..

I searched for "say number" in the * google archives and have not found reference to options for "say number". I would like to have * say digits instead of the hundreds and thousands. EG, "1234" would say one two three four.

Is there a way to clear the Queue stats? That is with out restarting *? I'd like to reset them daily and don't see a way to do that. Unless the only way is maybe a cron restart asterisk like every weekday @ 04:00?

Hi everyone! I was wondering... Does the musiconhold quality improve if the mpg123 processes run with a negative priority? If so, is there a way to make them start like that, so I don't have to renice them? Regards, Evert

> Message: 5 > Date: Fri, 12 Sep 2003 13:24:13 -0400 (EDT) > From: "David C. Troy" <dave@toad.net> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] 7206 as SIP->PSTN Gateway? > All, > > I know you can use, say, a 2620 w/2 port FXO card as > a SIP gateway. Clearly you can use the 5300, 5800, and > MGX8850 too. > >

Hello, I've search through all of the lists and cannot find any descriptions of live monitoring (monitoring a phone call going on between an extension and a zaptel channel live from another extension while the monitoring phone is muted). I am aware of the monitor function which is actually a call recorder, but I'm looking for live monitoring from a muted extension. is this easily

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

A friend and I have recently added the ability to announce the callers position in the call queue every x seconds.. or even just inject an anouncement every x seconds. All setup in queues.conf and can be setup per queue. My next project is to add the ability to announce the callers estimated wait time. I want some feedback to see whats the best method to calculate that? What do you want just

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

Hey folks -- Some of you had asked about getting my modification to ZapBarge that lets you monitor active Zap channels, scanning through them with the # key. I had posted a version a few days ago that was pretty crude; it didn't check to see which channels were active and assumed you had 23. I've finished a new version that checks to see what channels are actually active and pulls

Anybody have any experience pushing XML display data to a Cisco 7960? I am pulling data from a webserver now, but I have stuff, like queue status info, I'd love to be able to push at particular phones. A "pick up the damn phone" push message would be cool too. :) Dave ===================================================================== David C. Troy [dave@toad.net]

Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as "The White House" on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words "The White House" -- 202-456-1414 is an actual White House number. Using SetCIDName("Flying

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

All, I've been doing a lot of work on the Queue application and have some ideas I wanted to get some feedback on. Many people, myself included, have stated their desire for more/better queue statistics. We ought to be able to bring things up to a standard comparable to the established ACD apps. While some basic metrics (completed/abandoned calls, which agents last took a call, etc) can be

I have been transcoding a group of panoramic videos using ffmpeg2theora 0.26 i just completed transcoding Naples Cycle HDV.vwmovie the resulting video has some strange colors in it the source video can be got from http://www.eyesee360.com/videowarp/examples/ the resulting images and the theora file can be download from http://www.lightningridgecommunity.com/images/ tom at

On Fri, May 25, 2001 at 10:43:33AM +0200, Martin Hierling wrote: > HI, > > sounds like my problem discussed her two or three weeks ago. > Add "sleep_ratio 0" to your icecast.conf. > > regards Martin Thank you very muchly. It works now. Any idea how to synchronize 2 players on the same stream? End up with ~ 1 s phase difference. NAS might be more appropriate...

Hello all, A few days ago I noticed that the following seems to happen when upgradign OpenSSH on Linux-based (/etc/rc.d/init.d/sshd) based systems: - sshd is restarted with 'sshd restart'; however sshd serving in port 22 is not replaced. - you have to kill the old one (netstat -ltp | grep :ssh ; kill ...) first, then restart sshd. - This might happen only when performing the upgrade

Hello, I have a linux box with the following S/W versions: CentOS release 5.2 (Final) Linux ns.hostname.ro 2.6.18-92.1.13.el5 #1 SMP Wed Sep 24 19:32:05 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Version 3.0.28-1.el5_2.1 My server has 2 IP interfaces, eth0: with the public ip address and br0: for the private subnet. Br0 being a bridge ip interface between eth1 and tap0 ( for openvpn ). Now my

Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh -------------- next part