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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

...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 Administrador de la Red de Datos - UNET icamargo@unet.edu.ve >Message: 2 >From: "Kim C. Callis" <kcallis@c2associates.net> >To: <asterisk-users@lists.digium.com> >Date: Tue, 22 Jul 2003 20:09:49 -0700 >Organization: Callis, Chung & Associates, Inc. >Subject: [Asterisk-Users] QoS for Asterisk >Reply-To: asterisk-users@lists.digium.com > >This is a multi-part message in MIME format....

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 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:

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

I was finally playing around with the various databases, blacklist and cidname. The format for these are family name, key and value. Would that mean that the key is the name to enter, and the value the number? For instance: database put blacklist "Kim Callis" 3235551212? And if that is in fact correct, what happens if I don't add a key? -------------- next part -------------- An

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

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

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:

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

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.

I have come to realize that I don't have to have a g729a license in order to make use of an ATA-186 or 7460 connecting to another 7460. I just need to allow the codec in sip.conf. Now what ramification does that have when I dial out over one of my analog line (connected to * by a channelbank and a T100P) using my 7460 or ATA-186. The only benefit I am looking for is reduced bandwidth