Archives : September-2019
Im trying to track down a CPU spike we are seeing in a system.We have tracked down the spike to a single CPU and TID using that CPU.Indications are that its asterisk running this TID.Im trying to figure out what asterisk is doing on this thread aro..
How can I use an IF statement with a true value being a variable that has a colon in it?The colon in the true value variable is being taken as the delimiter for the false value.The only solution I came up with was some hackery to use STRREPLACE to repl..
all I maintain the above – it was set up by an external party with whom relations have now been severed by my employer. Quite early after the deployment it became evident that all .gsm audio files produced on this virtual instance at Azure via MixMoni..
Hallo, is there a Freepbx mailinglist? or can this be posted here?
Best Re..
does anyone know how i could use codec opus with asterisk 16 when using CentOS 6the prebuilt binary from digium doesnt loadThan..
All. I have an interesting scenario. We use the Polycom VXX phones and have an auto-attendant on our Asterisk system. The receptionist can turn the auto-attendant off and on as she would like (she dials 444 to enable and 555 to disable). However, Iā..
is there a way to view the call log (call history) from extern via browser, XML or whatever? At the moment I see no numbers in call-log. Asterisk do the CIDLokkup at the moment, and the Ring Groups has an CID Name Prefix. So an a Phone you can see: cal..
The Asterisk Development Team would like to announce security releases for Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1,15.7.4 and 16.5.1.These releases are available for immediate download athttps://downloads.asterisk.org/pub/telephony/asterisk/releases..
Thank you a lot for your kindly help and reply. Actually it helped me a lot.I was using _X. in the extensions.conf at the trunkinbound context.Can you advise me what is the difference between _X. and s? In other words, when it is better to use s ..
Hello;I am facing a trouble with the SIP service provider, they are saying that there is a problem related to message option 200 (the heartbeat), so what is required to add this for the sip configuration? Below is my sip debug trace log with the t..