similar to: Database usage?

Hi! Sure, just look for: Wonder Shaper. It's a HTB based shaper configuration wich have some very good features, I use a variation of that here at my College. http://lartc.org/wondershaper/ It is the page (a simple google search). Also make sure to uncomment the line tos=lowdelay in every config file of asterisk that have it. Hope it is usefull, sincerely, Ildefonso Camargo

Are there any other codecs that can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030723/34e950e0/attachment.htm

I noticed that there is a soft button for transfer when you initiate a call. I pressed it, and it actually put the call on hold, although I was able to call another extension. Is that soft button functional? And if so, how do you make use of it? And if not, how does one transfer a call? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I don't make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? Kim C. Callis

As I was working on my extensions.conf file, I started to segment calling privileges. For the everyday workers, I don't free reign to LD access unless it is business related. So I was wondering if there was a way to implement some type of accounting code to be entered before accessing LD, which of course would be noted in the CDR (however it is implemented, either comma delimited or MySQL).

I'm trying to figure out a way to make my own agent login, because I don't like how the default works. I have the login and logout working fine using the dynamic add and remove commands, but I need to be able to create a list of users and passwords. I thought of a way to do it using a list of passwords, but the agent would only ever be prompted for their password. I won't want that.

Yesterday, I started to experiment with Cisco to Cisco SIP calls using the g729 codec. According to the documentation, both the ATA-186 and 7960 are able to make use of the g729. >From an earlier e-mail, I made a change to the configuration of the ATA, changing the values: LBRCodec:3 RxCodec: 3 TxCodec: 3 The first thing I noticed was that when I did a sip show channels, the format had

Alright, I am basically cheap, and I have a cellular plan which allows for free incoming calls (Nextel). I was wondering if there was any way to do sort of a dialback trick in the extensions.conf. I call into the system from my cell phone (maybe via DISA), I dial an internal extension, and dial a phone number. Then * sends to my cellphone the number dialed thus giving me a in call on the cell. Or

Considering that I had a failure with compiling the latest version of asterisk because of cdr_mysql, I am going to assume that I need to have a copy of the mysql headers on the system in order to compile cdr_mysql.so. Does that sound correct? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've recently come across LDAPget (version 2.0rc1) and I've been trying to get it functional in my test environment (Asterisk 1.4.15 and MS Active Directory 2003) but I can't seem to get it working. I put together a test extension to try to change the CALLERID(name) by way of a LDAP query to AD: extensions.conf exten => 100,1,Answer() exten =>

I was trying to do a little searching to see if there has even been a comparison between Asterisk and VOCAL or any of the other OSS packages? "Practical Voice Over IP using VOCAL" published by O'Reilly and Associates, attempts to make a strong case about how scalable VOCAL. Of course, considering that the book is written by the makers of VOCAL, it tends to have a one sided slant.

Using * 1.0.9 I have a cron job that runs every night and sucks Caller ID information from our SQL server based CRM and imports it into the Asterisk database. We use the Caller ID to give enhanced information about the caller ("this is his customer number, he's a pain in the ass", for example). There is one guy in our database with no phone number and he has been sucked in to the

Hi All, Asterisk 1.4.12 on CentOS 5 Sorry for a question that I'm guessing is obvious to most of you. I'm trying to revamp my dialplan. When I first created it, I had something like: exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => s,2,Dial(${rgMain},${RINGTIME},t) exten => s,3,VoiceMail(main at default) exten => s,103,VoiceMail(main at default)

My configuration for my CAC is 12FXO/12FXS. A few days ago, I was looking at my zapata.conf and started wondering if 1 maybe my configuration could be off. The following is my current zapata.conf: ; Configuration file ; Set initial values for all channels in the channel ; bank, and apply those settings to channels 13-24. ; ; This is "group 2", and is indicated by g2 in Dial ;

I have been using the ATA-186 with good success (with exception of the fact that you have to recycle it from time to time). The one thing I haven't been able to do is to figure out how to make use of parking, transfers, etc. My cordless doesn't seem to pass DTMF, so I haven't had any success with parking. As for transfers, I can do a flash and get dial tone, but then what are my

Several people have asked about integrating asterisk in a company that currently has a PBX installed. Considering the capital outlay for most PBXs, depending on the model and age of the system, most companies do not want to look at replacement. So, what should one look at in a PBX in order to integrate asterisk into a pre-existing PBX? I have read of some systems with T1 interfaces, but what

I have an Asterisk 1.4.2 system online and have built up quite a large blacklist of tele-spammers that have been calling us. Recently we swapped one of our DID numbers to a SIP provider that now prefixes all calls with +1xxxxxxxxxxxx in front of US numbers (we're in the USA) and I need to edit my blacklist database so that all numbers in there that don't have the +1 prefix in front

Problem: *Call* *waiting* comes in, I press flash to answer it, and the first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP. System: Analog stations and trunks running on a pair of TDM400's. It does NOT have * call* *waiting* from the phone company, and I have enabled it in all my conf files. The trunks are set to FXSKS and the stations are FXOKS. I am not using *call*

Hello, I'm trying to fill CID data into the astdb using AsteriskWin32's asterisk.exe, to no avail: The batch file stops after the first line, and just waits: ---------------------------------------- rem c:\cygroot\mystuff>import.bat rem rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 "My cellphone"' rem rem Asterisk module

Hi All, I'm looking for a way to filter the AstDB cidname family to show only those entries with a specified area code in the Asterisk CLI. If this were a SQL database it would be something like: SELECT number, name FROM cidname WHERE number LIKE '1234%' I've tried "database show cidname 1234*" and substituted "%", "$", "-" for the wildcard