similar to: Transfers on the Cisco 7960

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

Is the CDR table the right table for billing? I did some tests and CDR records billing seconds for calls that where never picked up. Is this a bug in my system or is that the way CDR works? I called out on my X100T card. Best regards, Han Test data Duration 12 seconds 8 seconds billing time (never picked up my phone) Duration 111 seconds 108 seconds billing time (5 second but

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

Is there a way to get Music-on-Hold to use the Line input of the soundcard instead of playing MP3s? Some potential clients like the music-on-hold to be the local radio station, or from a satellite broadcast. Thanks!

Hi, The PBX at the Colorado Organization for Victims' Assistance fried as a result of the building power being cycled. I'm now in the process of building an * system to replace the failed PBX. Minimum cost is the priority. I have a T100P card installed in the new system, and I am about to order integrated T1 services from the "CBeyond" company. They will require eight

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

Arrrrgh I hate trying to sift through all these messages and keep track of the various threads going on ......... Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers....... only problem being that this list requires list membership before

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

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).

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

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.

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:

Hello, I have been wrestling with installing the CAPI drivers for AVM Fritz in order to use chan_capi with Asterisk. I use an SMP machine, RH 9. I have found rpm's for CAPI and AVM drivers (namely: capi4k-utils-2003.06.16-08.mungo.RH9.i686.rpm and kernel-2.4.20-8- avmfcpci-03.11.02-08.mungo.RH9.i686.rpm), but I believe that they support only single processor machines. I've already

hello i am trying to make a callback solution. client will call callback number and call is terminated. now callback server will create a call for that client. actually i have a problem in this process. that server is creating call to client (UA) when previous call is not disconnected yet. UA---------->Asterisk(callbacknumber) callis answered UA<----------Asterisk(callbackserver) call is

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

As of today many Travis users are seeing this error: The following packages have unmet dependencies: libmagick++-dev : Depends: libmagickcore-dev (= 8:6.7.7.10-6ubuntu3.9) but it is not going to be installed Depends: libmagickwand-dev (= 8:6.7.7.10-6ubuntu3.9) but it is not going to be installed E: Unable to correct problems, you have held broken packages. The problem is

Ok.. I got * and SIP working internally now .. still wrestling with connecting two remote * pbx's together.. I'll save that for another day though :) I setup Meetme on this new * PBX, but when I try to dial to join the conference, I hear a recording saying the conference is invalid or something to that effect. Here's a copy of my log files: == Parsing

Hi, I'm hoping someone has a suggestion for us. We have an ITSP that sends inbound traffic to us. Unannounced to us last week they started alternately sending traffic from two IP addresses, instead of the one we knew about. Some calls would pass, and others would be dumped as unauthenticated. I added the 2nd IP to the sip.conf file to allow for this, and everything was fine

Hello, POP users often keep fetched mail on server, by checking for "leave mail on server" or similar option in their client. What would be the best way to disallow them using such approach. That is, what is the best way to force deletion of messages that have been RETRieved in a POP session at the end of that POP session? RETRieved messages will be marked as \Seen, but since users

There is an unannounced and non-backwards-compatible change to the behaviour of 'get' in R2.8.0. 'get'ting a missing value now causes an error, whereas hitherto it's just returned a "missing" object. For example, in R2.8.0 this happens: test> getto <- function( x) get( 'x', sys.frame(1)) test> getto() Error in get("x", sys.frame(1)) :