[asterisk-users] trixbox

Hi Does trixbox comes with a predictive dialer, i want to use a predictive dialer with trix box or asterisk, please let me know what is the best tot use. Regards Kanishka ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailin

[asterisk-users] Install Asterisk on VPS

has any one tried installing asterisk on a VPS mechine ?   what is the minimum RAM and hard disk space needed to install asterisk if i am going to install it on a VPS mechine ?     ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-

[Asterisk-Users] asterisk on AMD 64 BIT

Hey Does asterisk works well on an AMD 64 bit processor server. are there any issues with this ? Regards Kani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://list

[Asterisk-Users] Web based interface

hello is there a web based interface for IVR management, check voice mail, check recorded calls and ect. regards kani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Asterisk on amd SERVER

Hi I am going to install asterisk on an AMD server, did any one had problems installing it on an AMD server ? Regards Kani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visi

[Asterisk-Users] class 5 softphone

hi guys what is a class 5 soft phone, i did a search on google, didn;t find, please let me know if any one knows. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Sip Behind Proxy

Hi I have a proxy server running and i want to have a sipura IP phone behind it. it does not work, but it works when it's behind nat, not proxy. is there a place in Ip phones to give a proxy address. please help me to configure this. Regards Kani

[Asterisk-Users] Asterisk SIP PORTS

Hi I am running asterisk SIP on port 5060, in my sipura i changed the 5060 port to 6060. but it's still tring to register it to asterisk. how come this is possible,   Regards Kani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-U

[Asterisk-Users] LD_LIBRARY_PATH

HiI set the LD_LIBRARY_PATH and when i reboot i have to set it again. how do i set it in linux to load it when the server reboots.RegardsKani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or u

[Asterisk-Users] No of records in calls table

Hi I use asterisk and Alepo, i got 20 mil records in the calls table and like 60 mil records in the failed calls table. it make the system very slow. how many records can a database handle normally.   how many records do you'll have in ur dbs   Regards Kani ___

[Asterisk-Users] asterisk and h323 problems

Hi i managed to install asterisk and h323, i am facing few problems, please help me 1.) i have setup the LD_LIBRARY_PATH like the following, but i have set it again when i reboot the server, how to slove this issue. PWLIBDIR=$HOME/pwlib export PWLIBDIR OPENH323DIR=$HOME/openh323 export OPENH323DI

[Asterisk-Users] asterisk + H323 + 723

Hi I am using asterisk 1.2.1, does any one has any luck with asterisk and h323. I want to use the codecs 723 and 729 with it. I am having one way audio issues with oh323 with I receive a call to asterieks through 723 . is there a successful implementation ? regards kani __

[Asterisk-Users] asterisk + H323 + 723

Hi I am using asterisk 1.2.1, does any one has any luck with asterisk and h323. I want to use the codecs 723 and 729 with it. I am having one way audio issues with oh323 with I receive a call to asterieks through 723 . is there a successful implementation ? regards kani ___

[Asterisk-Users] Asterisk 1.2

Hi Is there a release date for asterisk 1.2. I thought it'll be released this month.   can we upgrade from asterisk 1.0.9 or have to do a fresh installation once it's released.   tks kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- A

[Asterisk-Users] dropping extra frame of G.729 since we already have a VAD frame at the end

Hi I get the following error when i make a call from 729 to 729 dropping extra frame of G.729 since we already have a VAD frame at the end I am using asterisk 1.0.9, there was a patch for CVS vertion, is there a patch for 1.0.9 tks kani ___ --Bandwi

[Asterisk-Users] Re: Re: Sipura SPA 2000 - error using second line

Kanishka Somaratne wrote:> Hi> I have a Sipura SPA 2000 unit and I have configured both the lines in the> unit. both the lines are configured to use 729.> > when I make calls from the lines independently it works great. no > problem at> all.> > when line 1 is conne

[Asterisk-Users] Prblem with 723 and 729

Hi I have G729 and G723 codecs installed, I made some calling using a SIP IP phone. when I used the codecs 723 and 729 the call volume is less and the sound is little jerky, it's like call signals coming in and out. when I use gsm or G711 it works great sound quality is crystal clear. is this

[Asterisk-Users] Asterisk CDR

Where does asterisk store the CDR information by default, just after a fresh instalation. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Sipura SPA 2000 - error using second line

HiI have a Sipura SPA 2000 unit and I have configured both the lines in the unit. both the lines are configured to use 729.when I make calls from the lines independently it works great. no problem at all.when line 1 is connected and when I try to make a call using line 2 while line 1 is con

[Asterisk-Users] Problem With Sipura

Hi I have a Sipura SPA 2000 unit and I have configured both the lines in the unit. both the lines are configured to use 729. when I make calls from the lines independently it works great. no problem at all. when line 1 is connected and when I try to make a call using line 2 while line 1 is

[Asterisk-Users] ASTBILL

hi can we install astbill under mysql 4, or is mysql 5 a must regards kanishka ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-use

[Asterisk-Users] areski Problem

HiI have a problem with areski, if i am logged ni to the admin control panel and then at the same time if i login to a customers control panel. then it shows me there CDR information.If i login to the customer control panel directly it logs in but does not show the CDR. please check this the

[Asterisk-Users] Asterisk Billing

Hi I am looking for a asterisk billing system with a reseller module. for example, i there are 2 accoutns admin 1 and admin 2. when they login only the accounts they created should be shown. admin 2s accounts pr rates should not be shown to admin 2. does astbill support this. please let me kno

[Asterisk-Users] Problems Calling PSTN PSTN FROM ASTERISK

Hi I terminated a call through SIP to a landphone i have the following problems. 1.) asterisk gives a fake riming tone, it does not give the real tone from the phone company. 2.) when I put the call on hold the on hold music is not very clear. but when I talk the call quality is very clear.

[Asterisk-Users] No of simultaneous calls in asterisk

Hi Guys I want to know some details about the limits of my asterisk server Server Configurations is as bellow PIV 2.4 1GB RAM RedHAT LINUX 8 If any one know please let me know the following 1. ) how many simultaneous calls can asterisk handle with the server in pass - thru mode 2.) how many s

[Asterisk-Users] Bandwidth usage for codecs

hi how much bandwidth is used for the following codecs 723 r 5.3 723 r 6.3 723 r 8 what i know so far is the 723 r 5.3 uses 5.3 k up and 5.3k down 723 r 6.3 uses 6.3 k up and 6.3k down 729 r 8 uses 8 k up and 8k down is this correct or is it like the following 723 r 5.3 uses 11 k up and

[Asterisk-Users] asterisk-oh323-0.6.7

there is a problem with oh323 and incomming calls . the problem is at https://skylab.inaccessnetworks.com/mantis/view.php?id=15 there is a patch to solve this issue, has any one used the patch with oh323-0.6.7 ___ --Bandwidth and Colocation sponsore

[Asterisk-Users] H323 and Asterisk

hi guys I was working on asterisk and h323 for the past 2 weeks i have the following feedback please let me know if i am wrong h323 implementation I managed to install this it works, but the problem is it accecpts all calls from all ips. there is no way i can let it accecpt calls only from the I

[Asterisk-Users] OOH323C

hi has any one used OOH323C i tried this it is installed but do not know how to configure has any one used this, what is the best h323 addon to use with asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing lis

[Asterisk-Users] Asterisk on windows

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.co

[Asterisk-Users] Sip clients through proxy

Hi i know that we can use sip clients through nat, like the same way can we use sip clients through a proxy,. is there any sip client that i can specify a proxy address and use or any sip device. regards Kanishka ___ --Bandwidth and Colocation spo

[Asterisk-Users] problem calling SIP accounts

Hi I have configured sip accounts and they work some times. when i make a call to another SIP account it works right but some times i get the following error Jul 29 07:17:00 WARNING[802]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Re

[Asterisk-Users] CLI Numbers

Hi guys I have an asterisk server running, when some one make a call using asterisk i want to send a random CLI number from a CLI number list i have as the CLi number. how do i do this. i do not mind paying some one who can do this for me.   asterisk should not send the original CLi number o

[Asterisk-Users] astcc

HI I installed ACTCC, when i enter the pin number it says this call will cost 4.04 cents, it does not give a message like you have 100 mins. how do i get a message about the no of mins i have   Tks Kanishka ___ Asterisk-Users mailing list Asterisk-Us

[Asterisk-Users] reply a post

Hi how do i reply a question asked in this mailling list.   tks Kanishka   ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.

[Asterisk-Users] music on hold error

I have installed asterisk 1.0.6 i am using xlite for testing.when i transfer a call i get the music on hold when i put a user on hold using Xlite i get no sound at all. dead airwhy is that, in the asterisk log it is not even tring to paly the music on holdI have me extention like the fo

[Asterisk-Users] oh323 and open 729

has any one installed this, i just tried this on a test server, i get voice but it's corrupted, i do not get the natural voice any idea why ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster

[Asterisk-Users] Accecpt SIP calls from an IP

Hi I want to enable SIP calls from an ip address, direct calling without registering, the ip which sends the calls will not change. i have the following in the sip.conf file [cisco4] type=friend host=192.168.0.5; This device registers with us canreinvite=no ; Asterisk by default tries to redirec

[Asterisk-Users] Asterisk RealTime

Hi To install asterisk realtime we have to get the asterisk from CVS, is this stable and good to use ? any bugs ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE o

[Asterisk-Users] ASTCC Functions

Does ASTCC has functions like press a button and topup another card before it runs out of credit and check the balance which talking (by pressing a * 8 or some number) or if i make a mistake while entering the pin press ## and re enter.   is there a place where i can find all the key pad fun

[Asterisk-Users] ASTCC

Hi I installed ASTCC and got it working, when i enter the pin number and dialled the number needed, it says this call will cost point 20 cents per minute, can i get a message like you have 40 minutes and 30 seconds than giving the per min rate ?   Thank You Kani __

[Asterisk-Users] Asterisk Billing System

Hi Is there a billing system that i can view all the call taken by SIP clients in asterisk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Web based tool asterisk real time

Is there a webbased tool to use with asterisk real time. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailma

[Asterisk-Users] Unable to create channel of type 'SIP'

Hi I get the following error when i dial a sip extension, please help    NOTICE[1681]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP'  == Everyone is busy/congested at this time ___ Asterisk-Users mailing list Asterisk-Users@lists

[Asterisk-Users] Asterisk SIP client problem

Hi I have asterisk running on a server out side the office, this is with real ip. i have 1 realip to office and we share internet through nat. i have 5 SIP clients registered to asterisk from behind nat. when one of the sip cleints dial another sip clients extention the call does not come. when

[Asterisk-Users] best calling card platform for asterisk

what is the best calling card platform for asterisk ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/li

[Asterisk-Users] asterisk 1.0.5

can i install this directly or do i have to install 1.0.0 and then upgrade ?   ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://li

[Asterisk-Users] SIP broadband phone addon for asterisk

Hi Is there a add-on for asterisk where I can define a rate plan for outgoing international calls and let my sip users make calls depending on the credit they have.   tks Kanishka ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http:/

[Asterisk-Users] limit SIP extention outgoing calls

Hi how do i set an SIP users to make outgoing calls that is worth only $5. if they exceed $5 they can't make any calls. what i need is not a calling card, but  to limit outgoing calls for SIP users depedning on a value i give.   I use realtime asterisk.   Thank You Kanishka  _

[Asterisk-Users] open 723

has any one implemented open 723 at http://www.readytechnology.co.uk/open/g723.1   ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http:

[Asterisk-Users] Asterisk and 723,729

has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ??   ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/ast

[Asterisk-Users] do i have to reload asterisk every thing i add a new extension

do i have to reload asterisk every thing i add a new extension ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailma

[Asterisk-Users] do i have to reload asterisk every thing i add a neww extension

1.) do I have to keep reloading asterisk every thing I add a new extension or a new SIP user. 2.) is there a way to get the SIP users and the Extensions from a database. 3.) what can i do if i connect asterisk to mysql. i have seen this in voip-info site. __

[Asterisk-Users] Asterisk manager

What is the best Asterisk manager to use, i do not mind web based or GUI. Thank You Kanishka ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Send outgoing calls to a SIP gateway

How do I route all the outgoing calls through a SIP gateway, this should send more than one outgoing call to the sip gateway at once. please show me the sample configurations on how to do this.   my SIP gatway can accecpt direct IP traffic or SIP proxy traffc.   Thank You Kanishka  

[Asterisk-Users] Creating extension groups

Hi I want to create 2 groups of extensions, for example group 1 can’t make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server.   Please let me know how to do this. I was go

[Asterisk-Users] Asterisk manager

What is the best Asterisk manager to use, i do not mind web based or GUI.   Thank You Kanishka ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options vi