similar to: how to install asterisk

My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,

I have an asterisk box using a SIP provider and IAX2 softphones clients. Audio is low and I need to apply some gain on it. How can I configure such gains - in/out on sip and iax2 channels? Thanks, Valter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101106/85417a17/attachment.htm

We redirect calls to another pbx - which are not under our control. So, we need to distinguish when an operator answer from the time the call stays on queue. To do so, we use dial cmd with M option. M calls a macro that repeats a digit (saydigit) until someone dial it. When operator dials the correct digit we finish the macro making it to be bridged (caller and called). It almost works

Hi ! I've just finished setting up a samba 2.2.6pre2 installation on a SGI running IRIX 6.5.16. I've discovered that logging on a win2000 client, if I create a new file on a test share everything works but when I try to create a new directory I get the following error on win: Impossible to create the folder "New Folder" Impossible to find the specified file. Increasing the log

hi friends, i have installed asterisk,libpri,dahdi tar files in /usr/src. problem is that when i compile (./configure) asterisk-1.4.26.3, gtk+2.0.0 dependency is missing. i installed gtk from gtk.org, now when i am compiling gtk (./configure), i am getting this error message configure: WARNING: *** TIFF plug-in will not be built (TIFF library not found) *** configure: error: *** Checks for TIFF

Please stop emailing me personally. If no one replies to a post, it means that everyone is busy or they think you should read through the documentation before posting. If you can't figure out simple things like Music on hold from the documentation, then i dont think VOIP is for you. -----Original Message----- From: asterisk at opensourcesolution.in [mailto:asterisk at opensourcesolution.in]

I'm looking at version.h installed by Asterisk 1.2.3/4 - and the default value is 000000. I thought the value should be 010200. I know many people have problems compiling chan_bluetooth because of this inconsistency. Anyone has the last word on this?

i have installed database POSTGRESQL for storing call details. when i restart database i get the error. [root at localhost server]# psql -h 127.0.0.1 -U asterisk Password psql: could not connect to server: Connection refused Is the server running on host "127.0.0.1" and accepting TCP/IP connections on port 5432? THIS IS MY /VAR/LIB/PGSQL/DATA/POSTGRESQL.CONF # CONNECTIONS AND

hi friends, as i am a beginner in voip, i had made a very simple dial plan i had made two extentions n both are able to ring each other through soft phone (X-Lite) below is my dialplan ################################################################################################################################### /ETC/EXTENSIONS.CONF [others] [my-phones] exten =>

As I said, please keep discussion on list. asterisk at opensourcesolution.in wrote: > hi all, > > first of all i appologise for sending u pvt email. i have installed > asterisk on Centos 5.3, plz open the attachment in which i had drawn a > tolpology. i had installed one asterisk machine and two windows machine. > now i want to install softphone in both windows machine.

hi friends, i had installed postgres database for call log,call detail. it has restarted succesfully but when i check tcp connection i dont get any welcome message by psql. [root at localhost ~]# # psql -h 127.0.0.1 -U asterisk password [root at localhost ~]# th in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Not able to compile asterisk,zaptel,libpri in /usr/src -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100208/fa5ff126/attachment.htm

hello friends friend i had just finished my chapters of asterisk. ill be configuring asterisk in for home for r/d purpose. i am having p4 machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the only doubt which i am having is which hardware ill have to buy to configure asterisk. i think analog card ? plz clear my doubt. n be with me from beginning till end, of the journey of

Hi, On my Suse90-out of box I had downloaded from CVS asterisk. I'm running kernel 2.4.21-99-smp4g with 4cpu's, and the kernelsource is in /usr/src/linux Asterisk compiles with no problem. But when compiling zaptel I got this error .......... ........ ........ zaptel.c: In function `zt_ec_chunk': zaptel.c:4981: error: parse error before "unsigned" zaptel.c: In function

hello all, friends i am new in asterisk. i had just finished the installation requirment of asterisk. i am using Centos 5.3 in which ill be installing asterisk now guys plz guide me my requirment for deploying asterisk is, i am having a client, (HR Consultancy) where 40 executives work and on each 40 desk, phone is there. i want confrencing facility,hold facility,extention nos,music. when ever

Hi all, my domU guests stop booting in my new 3.2 (kernel 2.6.18.8) xen installation when run perfectly in my other 3.0.2 (2.6.16) xen server. whe i start the domU with "xm create" i get these messages: Using config file "./debian.4-0.xen3.cfg". Started domain debian.4-0 IP route cache hash table entries: 4096 (order: 2, 16384 bytes) TCP established hash table

Hi, Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such a way that I cannot script non-english sound files downloading anymore. The following used to work (unattended) with 1.6.1.9 (for instance): cd /usr/src/asterisk-${ASTERISK_VERSION} ./configure make menuselect.makeopts echo "MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM" >

I am still getting an error compiling the 1.2-Beta version. The tarball works fine, but I have never been able to compile the 1.2beta from CVS. I have been compiling CVS-HEAD on the machine for quite some time. It goes into this loop: if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \

Hi All, Is it possible to establish site-to-site VPN using dynamic IP addresss assigned by ISP ? If yes, I would like to request a sample ipsec.conf for such scenario... Thanks and warmest regards aslay ################################################### # This message has been scanned for viruses and # # dangerous content by Pensteel Digital Solutions # # Open Source Security Server,

Due to a packaging error, the tarball released on Friday night did not have a version number embedded in it, which results in various strange build errors and other odd behavior. The tarball on the FTP servers has been updated to correct this situation. Sorry for the inconvenience :-)