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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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050223/6f9ff4c3/attachment.htm

...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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050223/cd32fa4b/attachment.htm

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, As you know, we released Asterisk 1.0.4 earlier this week. However, there was a callerid bug in chan_zap that has caused us to go ahead and make another release. Asterisk 1.0.5 is available at all of the usual locations. I'm sorry for any inconvenience this may cause. Russell Bryant -----BEGIN PGP SIGNATURE----- Version: GnuPG

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050227/73c38e9c/attachment.htm

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050315/af5a3bb2/attachment.htm

Hi I am going to install asterisk on an AMD server, did any one had problems installing it on an AMD server ? Regards Kani

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF

...ection seems to be still established giving provision for the third client to connect to the server. As this behaviour continues the number of users whom can be connected get reduced because of these connections still being established. (ie MaxStartups - set as 3). In syslog.log: Jan 27 03:49:58 kanishka sshd[7056]: fatal: Timeout before authentication for 127.0.0.1 Jan 27 03:49:59 kanishka sshd[7075]: invalid module type: configuration Example of netstat -an|grep 22: tcp 0 0 127.0.0.1.22 127.0.0.1.58651 ESTABLISHED tcp 0 0 127.0.0.1.22 127.0.0.1.58647 ESTABLISH...

has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050225/d5daf369/attachment.htm

I've got the same problem. MusicOnHold works if I use something like: Exten => 1111,1,MusicOnHold() but if I try to answer a call and then transfer or put on hold the call, I get no music. Does anyone have any idea? Bye, Gianluca. _____ Da: Kanishka Somaratne [mailto:kani@technoportal.biz] Inviato: gioved? 17 marzo 2005 5.53 A: asterisk-users@lists.digium.com Oggetto: [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 u...

...sip UA to h323 gateway, can't get Call-id pass from sip UA to h323 gateway, h323 always gets call-ID sent from Asterisk as *. are there any configure to pass the correct call-id from sip UA to h323 gateway? or this is a bug in oh323 0.67? how about oh323 0.73 ? Mario On 9/29/05, Kanishka Somaratne <kani@technoportal.biz> wrote: > 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 call...

why can't we compile the asterisk coading in windows, it's done in c++ so it should work in windows as well

...them will do no good. Connect to the CAC and ask it for the T1 configuration to verify what it really is. Also make sure you rerun ztcfg after any changes to zaptel.conf. What does zttool tell you? ------------------------------ Message: 17 Date: Fri, 18 Mar 2005 16:15:24 -0000 From: "Kanishka Somaratne" <kani@technoportal.biz> Subject: [Asterisk-Users] reply a post To: <asterisk-users@lists.digium.com> Message-ID: <005c01c52bd5$b2ba2820$0200a8c0@CYBER1> Content-Type: text/plain; charset="iso-8859-1" Hi how do i reply a question asked in this mailling li...

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