similar to: * Configuration

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!

Hi, I have received some replies for my previous mail (* configuration), asking for my goals in configuring Asterisk. So here they are: We are planning to host an Inter continental virtual PBX service that will enable our users to register for an account which give them a toll-free # or a DID. Once registered using a web based interface, that user can add as manay extensions he/she wants

Hello, >From: "Hermann Wecke" <hermann@wecke.com> >Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern => >extensions.conf >Date: 8 May 2004 22:03:57 +0000 > >Is it possible to strip some numbers from the *end* of a number? > >I know that ${EXTEN:1} will remove 1 position from the beggining... but >how to remove N numbers from

Hi there, I would like to communicate H323 IP phones with SIP phones. My H323 phones are registered to a gnugk GK, and the SIP phones are registered to a asterisk SIP proxy. I could not create a dialplan that works. Inside my extensions.conf file I created the following two entrances: exten => 4,1,Dial(SIP/4) exten => 5,1,Dial(SIP/5) This allows SIP phones call each other.

I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening: localhost*CLI> -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtest May 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019

Hi. I'm trying to start a MGCP configuration in Asterisk but i have some basic problems. I hope that someone can help me. First ..how do set two call agents in the configuration files? How is the extensions.conf for MGCP?! I'm trying to start the Asterisk, and obtain this: [root@server3 asterisk]# ./asterisk -vvvc == Parsing '/etc/asterisk/asterisk.conf': Not found (No such

Hi all, I am trying to write a program that sends SIP requests to asterisk. My aim is to make asterisk record as many voicemails it can at a time. The design of the program is like this: There are two processes: One main process and a child process (No flames pls. I have very little idea about pthreads and dl modules) The main program asks the user to input the number of test instances. When

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

Hi friends, I have the entry exten => h,Hangup in my extensions.conf, and I am trying to record the call details for billing. From the wiki i found out that the use of "exten=>h,..." is not suggested for the CDRs. What impact will the use of 'h' make on CDRs? Also, what is the advantage of using ResetCDR with exten=>h? Regards... Girish

<DIV><FONT face=Arial size=2>hello All,</FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9</FONT></DIV> <DIV><FONT face=Arial size=2>to take an outside call through the hardware pbx, our

I have the standard parking.conf but extension 700 doesn't show up in my dialplan.... Why? I can dial 701 which tells me that I don't have any calls parked there. 700 just gives me invalid extension noise.... Should I have extension 700 defined elsewhere? Thanks parking.conf [general] parkext =a 700 ; What ext. to dial to park parkpos => 701-705

We're getting ready to ditch our hosted virtual PBX for an Asterisk solution and I've got a couple questions that probably come from experience; we're looking to host two PBXs on the same box if the PBXs are identified by different DID numbers, and I was curious if: 1. Is it possible to have duplicate extensions between the two PBXs? Eg: 555-1111x100 and 555-2222x100 on same

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of

HI all, i am configuring VNC server in CentOS 5.1. i have enables this two lines in /etc/sysconfig/vncserver file VNCSERVERS="2:vnc" VNCSERVERARGS[2]="-geometry 1024x768 -depth 24" here i have created a user vnc and set the password using vncpasswd cmd. then i started the vncserver # vncserver New 'localhost.localdomain:1 (gopinath)' desktop is

How to configure Diskless environment in CentOS 5.1. Please help me out Regards, Gopinath M Signal Networks Pvt. Ltd. Smile... it increases your face value! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20080410/bf9d216c/attachment-0001.html>

hi, how to write a scripts which launches 10 pings to different destinations at execution of single shell scripts please help me any ideas regards, Gopinath

hi, how to upload and download file from a ftp server using GUI. ftp client. upload and download to and from ftp server works fine when done done from command prompt but not working from GUI ftp client Ex. Gftp. is there any modifications to be done on the server, regards gopinath -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi i am new to linux while installing mplayer [gopinath at localhost Desktop]$ rpm -ivh mplayer-1.0-0.37.rc1try2.el5.rf.i386.rpm mplayer-fonts-1.1-3.0.rf.noarch.rpm warning: mplayer-1.0-0.37.rc1try2.el5.rf.i386.rpm: Header V3 DSA signature: NOKEY, key ID 6b8d79e6 error: Failed dependencies: libaa.so.1 is needed by mplayer-1.0-0.37.rc1try2.el5.rf.i386 libcaca.so.0 is needed by

Hi Experts, I am using icecast to stream the audio from asterisk server. we using asterisk voip server for emergency announcement for user extensions which are associated with that server. During emergency announcement if any one of user is offline or their application running in background he will not listen announcement. To overcome this, we are streaming the same announcement to icecast and

dear all i have a web server on centos 4.6. i have configured one more back up Webserver with the same configuration as the same on the primary web server Now how to configure this servers for fault tolerance. that is. if one server gets down the other should automatically become up. Please any one help me out how to configure the above. i am new to Linux ... Regards, Gopinath M Signal