similar to: ASTERISK AND 120 CONCURRENT CALLS

Just wondering if anyone has tried MAC OS X and panther. I will like to do SIP to H323, not sure if this will be possible on the MAC because of the Libraries PWlib and OPenh32 for Linux.. Just curious.. Anyway, anyone has an easy guide (step by step) to setup oh323 with asterisk. I saw a guide but i am not very savy on linux. thanks, Francisco ----- Original Message ----- From:

Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks

Hi, does anybody manage to get music-on-hold with inaccess oh323 driver? Statement like : exten => 10,1,Dial(OH323/xx,mt) works (dials the xx number) but no music is heared. Also, if I put 'r' (ringback) it doesn't work either. With chan_h323 I got this functionality but this driver had some other problems (call transfer don't work).... Thanx in advance, Victor...

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

hi for all, someone have any experiencie with dovecot and postfix with 200 users in /etc/passwd as local user? is a funcionatly, or have some errors in the future?, and how the users can change they passwords?, now for change passowrd I have to use the #passwd usuerx and to add a user # adduser, the populars commando, please some recomendation? -------------- next part -------------- An HTML

Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. we are planning on using 4 servers to serve the calls and one for the database servers' specifications are: 2.8 Dual Core Proccessors 2 GB Ram 160 Sata Drive each server will be provided with 1 E1 card Questions are: 1- will those servers be able to handle that ammount

Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _________________________________________________________________ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now!

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK

is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working????? I am using chan_h323 Thanks!! Sebastian Nocetti. --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date:

Hello, Well, i need to limit the number of elements in a sortable. I new to scriptaculo but i have lots of experiencie with javascript. Since limit the number of elements is not supported by Sortable i would like to implement, i already took a look at the Sortable class in dragdrop.js and it looks like that onHover and onEmptyHover deal with the update of the UI, however I''m not unable

Sorry for reposting, i keep forgetting this should be plain text. Will not make this mistake again Hello, The following code moo <- function(a=1){ x=1; x=x+y} funs <- new.env() enter <- function(type, v, e, w){ assign(v, TRUE, funs) } library(codetools) collectUsage(moo, enterGlobal = enter) adds + to the environment funs i.e. funs: "=" "{" "+"

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

Does Asterisk support Radius accounting?.... -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send Asterisk-Users mailing list

Hello, This is not really a shorewall problem. But just wanted to check if this problem rang a bell with any of you. I have a linux router with slackware 9.1, and kernel 2.4.27 Everyting works ok except for access to web sites that use akamai from behind the router. >From the router machine itself I can access those sites without problems. But machines behind nat, take forever to access

how many calls can be handled with Asterisk Calling Card app? somebody knows that? --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.744 / Virus Database: 496 - Release Date: 2004-08-24 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/498fd5f8/attachment.htm

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

Hello, Does anyone have an up-to-date page describing, step by step, how to make a customized CentOS5 DVD? I noticed that CentOS5 already comes with ~240MB of updates. So for starters, I'd like to create a new DVD with all the current updates. (And I have other custom scripts I need to install on top of that). I've googled around and tried various suggestions on the net:

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus -------------- next part -------------- An