I am running Asterisk 13.30.040 core CPU (VM) VMware. CentOS 732 G ram10G vmx networkShould be plenty of room for anything…Yes asterisk is running 270% CPU… Is it not taking advantage of the 40 cores ?I am bring around 300 SIP endpoints in a mu..
Author : Jerry Geis
Im trying to get a sense for how many video calls with the Confbridge can be active when dropping the incoming video with the confbridge setup.So its really just the main persons video is showing out to all the endpoints. So its a one to many kind..
What is the command to install dahdi on a systemctl type startup ?I just installed dahdi from git (so latest) and did :cd dahdi-linux-complete ls shows dahdi-linux and dahdi-tools find . | grep service shows nothing.in dahdi-tools there is the OLD dahdi.i..
How does an external program get notification of new registrations ?Would that come over the AMI or anything ?T..
So I have CentOS 7 server running asterisk 18.8.0 – all is good.I unplug that server – plug in a ubuntu 20.04 server at the same IP address. let my 3 devices reconnect to the ubuntu server….When I pick up the polycom phone and dial it connects. I h..
I am running 18.8.0 -videosupport is enabled. I get video calls no problem.However when I make a call file to a soft phone and include:Codecs: ulaw,h264in the call file…sip show channels – shows:4013c15f1f4cdff(ulaw|h264)No Tx: ACKso clearly the cal..
I am trying to run this command:exten => _4XX,n,System(/usr/bin/rm /tmp/test.incoming.txt)From the log: Executing [402@smvoice-sip:7] System(SIP/103-00000018, /usr/bin/rm/tmp/test.incoming.txt) in new stackIs rm not an allowed command – the above f..
– Any one using SIPML5 ? How many video connections can a normalasterisk serverbox (2.2G 8GIG ram) handle in a SINGLE video session ?Th..
I am trying to use the SIPML5 at https://www.doubango.org/sipml5/call.htm?svn%2and when I hit the login button – and asterisk says wrong password and the web page says Forbidden.I have triple checked that I entered the correct password on the websi..
-I have a device that has 16 RTP ports.I desire to bring that audio into Asterisk… is that possible ?The device does not run SIP at all just RTP audio. I am using Asterisk 18. How might I do that ?Th..