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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!

...nal 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 kphone 4.02, but when i'm make a call the programs >doesn't respond any command, so i can't hear any sound .. > > >in sip.conf that's my codec config: > >disallow=all...

hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing this....because we wanna implement it in some locations!! Thanks All!! Sebastian. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...two sip phones .. but when i try to make a call ( in the form : user@server) from the Kphone they returns : "Call Failed: Not Found" The extensions.conf is the default, only in sip.conf i add the peers .. what's wrong ? Regards, Ivan ________________________________________ FiberTel, el nombre de la banda ancha http://www.fibertel.com.ar

...tarters, 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: http://sipx-wiki.calivia.com/index.php/A_Kickstart_CD_for_sipX_on_CentOS http://cablemodem.fibertel.com.ar/lateral/stories/38.html However, I have not been able to find step by step instructions for CentOS5. For example, genhdlist has been deprecated. Also, the centos-release-5.0.0 rpm grabs a new GPG key from the net, even though an identical one is already on the DVD. This creates problems o...

...ccounts 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 up and running? SER/Asterisk on the same machine? rgds, /Staffan kerker -----Ursprungligt meddelande----- Fr?n: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Skickat: den 6 november 2003 12:45 Till: asterisk-users@lists.digium.com ?mne: Re: [Asterisk-Users] How to control dialout in extensions file You could use DISA app. exten => 2101,1,DISA,/opt/pass.txt|default Where: /opt/pass.txt is a plain text file with password list. default is a...

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

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

Hi all: Somebody knows the mysql table structure for VoiceMail2 application? Thanks in advance, Gus

Hello, >From: klky3@fibertel.com.ar >To: asterisk-users@lists.digium.com >Subject: Asunto: Re: [Asterisk-Users] Syntax >Date: Wed, 5 May 2004 17:06:56 -0300 > >Somebody knows a Howto that have the examples, but with comments ( the >cookbok >in the digium's page is quite diffcultly !!! ) > >Best...

Hi !! I know that a conference room can be made infinitely. but, I think that there is actually a limit. For example, how many conference rooms can be made from CPU 866 [MHz] and RAM 256 [MB]? Is there any person who tried someone? I am studying MeetMe now. Please tell me a hint!!

Hi all: Very disappointed, finally I left the attended call transfer with ATA 186 using SIP. With image 2.16-1, ATA sens '486 - Busy Here' when trying to transfer the call.. I consulted with Cisco guys and accepts that some problems with this service exist. Soon as I can I will try using MGCP. My doubt now is if somebody proved the DynExtenDB application. I read some commentaries but

How do I use the Authenticate application in my IVR menu, where do I put the password? here is my menu. I need to ask for a password before I let users log into my conference room. [conf1] exten => s,1,Ringing exten => s,2,Wait,2 exten => s,3,Answer exten => s,4,Authenticate(1234) exten => s,5,Hangup exten => a,1,Meetme,1251 I also can not figure out what "Unknown RTP

Hi, If I have a PRI with all channels grouped in group=1, I understand when I want to make an outgoing call that asterisk takes the first channel available. Is there any possiblity to "rotate" the channel taken? I was searching in Wiki but I could not find nothing about. Thanks, Alejandro

...s.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 10 > From: "Sebastian Nocetti" <snocetti@fibertel.com.ar> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] STILL NO AUDIO > Date: Mon, 19 Jul 2004 12:51:49 -0300 > Reply-To: asterisk-users@lists.digium.com > > Testing both... > > -----Mensaje original----- > De: asterisk-users-admin@lists....

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 all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

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

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, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping