[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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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://lists.digium.com/mailman/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH 2.) when i send a call through h323 i get an error saying ouch broken audio pipe and asterisk stops, does any one have this problem please help ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 -- 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/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 connected and when I try to make a call using line 2 while> line 1 is connected I get codec error.> > what could be the problem , please help.> > I tried this with call the other codecs as well, i still get the same > error,> only when i am tring to make the second active call> > regards> kanishka>kanishka,>The SPA-2000 cannot support two simultaneous g729 calls. >You will need >to allow ulaw/alaw on both users (in sip.conf) in case it needs >to fall >back from g729.>>If you need two simultaneous g729 calls, the SPA-2100 will >support them.>-->Kristian Kielhofner Kristian thank you very much for the reply. what codecs does SPA-2000 support simultaneously. can it support 729 on line 1 and 723 on line 2. I tried this as well it failed. please let me know what codecs it support simultaneously. tks Kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 some thing to do with jitter buffer , is there a way to increase the volume using asterisk. tks kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 connected I get codec error.what could be the problem , please help.I tried this with call the other codecs as well, i still get the same error, only when i am tring to make the second active callregardskanishka ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 connected I get codec error. what could be the problem , please help. I tried this with call the other codecs as well, i still get the same error, only when i am tring to make the second active call 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-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 there is a problemit logs in show customer information and balance which is correct but not the CDRregardskani ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 know regards Kani ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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. if any of you guys have come across this please let me know what I did wrong. 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-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 simultaneous calls can asterisk handle with trancoding (with codec conversions) let me know if u guys know cheers Kanishka ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 11k down 723 r 6.3 uses 13 k up and 13k down 729 r 8 uses 16 k up and 16k down if u guy know, please let me know. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 IPs i give and bill depending on IP oh323 implementation managed to install, same as h323 implementation i can't add a list of ips and restrict access, the 729 -> 723.1 codec convertion does not work well, get a robort voice ooh323c installeed but do not know how to configure :( woomera let me know if there's any one who has tried this. what i want to do it accecpt h323 calls and bill depending on the ip address and send the calls via h323 depdning on the gateway IP i add Regards Kansihka ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 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/listinfo/asterisk-users
[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.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 sponsored by Easynews.com -- 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/listinfo/asterisk-users
[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 Request) this happence when i register the SIP users and stay for some time and dial.but no problem with out going calls, can call any time. Regards 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.digium.com/mailman/listinfo/asterisk-users
[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 out it must send a number from a list we have. (list can be a database or a txt file) 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.digium.com/mailman/listinfo/asterisk-users
[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-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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.digium.com/mailman/listinfo/asterisk-users
[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 followingexten => 2007,1,SetMusicOnHold(default)exten => 2007,2,Dial(SIP/xlite,30,Tt)what is the problem, ___ 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/listinfo/asterisk-users
[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/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 redirect the defaultip=192.168.0.5 context=income but this is not working, pls let me know if this is correct. 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.digium.com/mailman/listinfo/asterisk-users
[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 or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 functions for this. ___ 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/listinfo/asterisk-users
[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 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/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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/mailman/listinfo/asterisk-users
[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.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 some one who is out side the nat calls, the call comes. I also have nat = yes under SIPcleints. what did i do wrong. please help me. 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.digium.com/mailman/listinfo/asterisk-users
[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/listinfo/asterisk-users
[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://lists.digium.com/mailman/listinfo/asterisk-users
[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://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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/listinfo/asterisk-users
[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://lists.digium.com/mailman/listinfo/asterisk-users
[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/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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/mailman/listinfo/asterisk-users
[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 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/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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/listinfo/asterisk-users
[Asterisk-Users] Creating extension groups
Hi I want to create 2 groups of extensions, for example group 1 cant 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 going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please explain how to do this in detailed. I want to know how to route all outgoing calls through our SIP server and how to stop some of the extensions from taking outgoing calls 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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
