similar to: video confrencing

Hi I've a problem with a gnugkv2.0.7 I've compiled gnugk successfully I've installed PWlib-1.6.6 and openh323-1.13.5 libraries successfully When i run gnugk i have this error: error while loading shared libraries liboh323_linux_x86_r.so.1.13.5 cannot open shared object file No such file or directory I try to use command export: export

hello *'s, Astcc not workin what is correct format for defining 1-database 2-brands 3-trunks 4-routes i define all these things but not workin may be i define in wrong format.I have FXO card installed.can anyone implement it and also my sip phone generates very loud noise wat is that i tried several settings but not hear any voice just noise. sip.conf [general] context=from-sip port=5060

Hi Nicolas, There doesn't seem to be any interest in using asterisk and video. I posted a $1,000 bounty to get video meet me working without a single reply. I have now just bumped this to $2000 http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+Meet+Me+vid eo+conferencing This is a legitimate commercially binding bounty, I hope this might inspire some people to develop at least

Hello, We 'll setup asterisk servers on several remote locations atleast 6-7 different locations these are connected to each other through DXX(Digital Cross Connect) ,on larger locations we use PRI/E1 and small locations we use TDM400 may be one or two but lot of IP phones(soft/hard phones),basically we are currently in planning phase to which one is the best for implementing this setup

hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck my configs are. sip.conf [general] port=5060 bindaddr=192.168.10.195 disallow=all allow=alaw allow=ulaw [101] username=101 type=friend secret=1234 host=192.168.10.195 context=sip callerid="101"<101> defaultip=192.168.10.176 extensions.conf [globals] [incoming] exten =>

Hello *'s, I have recently installed CentOS v3.3 and i have latest stable Asterisk's source code ,i compiles it shows no error but when i am looking for sip.conf,zapata.conf ,i am amazed the /etc/asterisk folder was empty i am compling several times but no luck what's the problem i compiled in the order of zaptel,libpri,asterisk. I send some traces of my asterisk's compilation

Hello *'s, i am using Realtime Sip drivers but its not working here is my configs: extconfig.conf [settings] ; Realtime configuration engine ; ; maps a particular family of realtime ; configuration to a given database driver, ; database and table (or uses the name of ; the family if the table is not specified ; sipfriends => mysql,asterisk,sip_friends res_mysql.conf [general] dbhost =

Hi Mahdi, Did you get a chance to verify this fix again? If this fix works for you, is it OK if we move this bug to CLOSED state and revert the rebalance-cli warning patch? -Krutika On Mon, May 29, 2017 at 6:51 PM, Mahdi Adnan <mahdi.adnan at outlook.com> wrote: > Hello, > > > Yes, i forgot to upgrade the client as well. > > I did the upgrade and created a new volume,

OK. Is this problem unique to templates for a particular guest OS type? Or is this something you see for all guest OS? Also, can you get the output of `getfattr -d -m . -e hex <path>` for the following two "paths" from all of the bricks: path to the file representing the vm created off this template wrt the brick. It will usually be $BRICKPATH/xxxx....xx/images/$UUID where $UUID

Great, thanks! Il 5 giu 2017 6:49 AM, "Krutika Dhananjay" <kdhananj at redhat.com> ha scritto: > The fixes are already available in 3.10.2, 3.8.12 and 3.11.0 > > -Krutika > > On Sun, Jun 4, 2017 at 5:30 PM, Gandalf Corvotempesta < > gandalf.corvotempesta at gmail.com> wrote: > >> Great news. >> Is this planned to be published in next

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

Great news. Is this planned to be published in next release? Il 29 mag 2017 3:27 PM, "Krutika Dhananjay" <kdhananj at redhat.com> ha scritto: > Thanks for that update. Very happy to hear it ran fine without any issues. > :) > > Yeah so you can ignore those 'No such file or directory' errors. They > represent a transient state where DHT in the client process

The fixes are already available in 3.10.2, 3.8.12 and 3.11.0 -Krutika On Sun, Jun 4, 2017 at 5:30 PM, Gandalf Corvotempesta < gandalf.corvotempesta at gmail.com> wrote: > Great news. > Is this planned to be published in next release? > > Il 29 mag 2017 3:27 PM, "Krutika Dhananjay" <kdhananj at redhat.com> ha > scritto: > >> Thanks for that update.

Hi, Thank you for your reply. Lindsay, Uunfortunately i do not have backup for this template. Krutika, The stat-prefetch is already disabled on the volume. -- Respectfully Mahdi A. Mahdi ________________________________ From: Krutika Dhananjay <kdhananj at redhat.com> Sent: Friday, October 6, 2017 7:39 AM To: Lindsay Mathieson Cc: Mahdi Adnan; gluster-users at gluster.org Subject: Re:

hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck.I know very well this is not kind a problem discussed in this group but i try my best and all in vein so finally i am here hoping you ppl helping me out.I discussed this problem in asterisk's-users group and adding feedback from asterisk-users group my configs are sip.conf [general] port=5060

Hi ! I need a solution to park incoming calls to an extension of my choice where a special announcement is played, park subsequent calls to specific pools so that they listen to announcements of my choice. any ideas ? Shah.

Could you disable stat-prefetch on the volume and create another vm off that template and see if it works? -Krutika On Fri, Oct 6, 2017 at 8:28 AM, Lindsay Mathieson < lindsay.mathieson at gmail.com> wrote: > Any chance of a backup you could do bit compare with? > > > > Sent from my Windows 10 phone > > > > *From: *Mahdi Adnan <mahdi.adnan at outlook.com>

On Mon, Jul 10, 2017 at 10:33 PM, Mahdi Adnan <mahdi.adnan at outlook.com> wrote: > I upgraded from 3.8.12 to 3.8.13 without issues. > > Two replicated volumes with online update, upgraded clients first and > followed by servers upgrade, "stop glusterd, pkill gluster*, update > gluster*, start glusterd, monitor healing process and logs, after > completion proceed to

My goal is build a configuration that provides an 800 number to an internal (LAN) MCU Asterisk/OpenMCU server to manage an audio conference for up to 8 clients (for one hour) that will be streamed to an icecast2 server. I have googled and read a few discussions about the use of Asterisk and possiibly OpenMCU to stream audio to an icecast2 server. I have seen snippets here and there and

Hello ppl, At initial level i configure asterisk woth only soft phones ,in which one at windows machine and other is linux i am using windows messenger and linphone respectively both phones registered with asterisk respectively problem is that they bypass asterisk on call when i send request from linphone to messenger request shown on messenger but on asterisk console nothing to and also if i send