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...cledit iconv NLS profmem cairo TRUE FALSE TRUE TRUE TRUE TRUE Best regards, Ulrich -- ______________________________________________________________________ Ulrich Leopold Resource Centre for Environmental Technologies, Public Research Centre Henri Tudor, Technoport Schlassgoart, 66 rue de Luxembourg, P.O. BOX 144, L-4002 Esch-sur-Alzette, Luxembourg tel: +352 425991 618 fax: +352 425991 601 mobile: +352 691 304813 http://www.crte.lu Computational Bio- and Physical Geography, Institute for Biodiversity and Ecosystem Dynamics, University of Amsterdam, Nieuwe...

...uot;\\*",x,value=T))==1) return NULL instead of FALSE ! I you have any idea, please let me know ! Many thanks, Jessica ____________________________________ Jessica Gervais Mail: jessica.gervais at tudor.lu Resource Centre for Environmental Technologies, Public Research Centre Henri Tudor, Technoport Schlassgoart, 66 rue de Luxembourg, P.O. BOX 144, L-4002 Esch-sur-Alzette, Luxembourg (See attached file: test.txt) -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: test.txt Url: https://stat.ethz.ch/pipermail/r-help/attachments/20080212/b67d1c...

...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 using Xlite i get no sound at all....

...'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 calls > from all ips. there is...

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:

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

...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 list. tks Kanishka -------------- nex...

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

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:

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 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

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

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 going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please

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