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Ive noticed on several occasions that if Asterisk starts without a network connection, then even if the network connection is restored, DNS lookups fail. After the connection is restored I can successfully do NSLOOKUPs from the command line, but ..
everyone,Does anyone know is it possible to cancel the call duration limit set in app Dial with options S(x) or L(x[:y[:z]]), by for instance entering custom feature code (application map) during the call ?I have read somewhere that bridge features ..
I have the following scenario: Agent calls external number Mixmonitor starts recording call After agent speaks with customer they need to transfer them to an extension that will simply play a message Customer hangs up The problem is t..
,When using a SIP proxy to load balance calls how do you make it that a call on an attended transfer reaches the same Asterisk box every time? I was told that in later versions of Asterisk there is some magic to make it work correctly when load balancing..
Everybody,Ive recently discovered openai/whisper and have been trying in earnest to get this working with Asterisk for voicemail transcriptions (Currently using the NerdVittles script with IBM Watson)https://github.com/openai/whisperAfter spending seve..
What is maximum cps limit of a good asterisk server(single node ) ?regards*Tahir Almas*Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open sou..
in release notes for RHEL 9.1 i see — https://access.redhat.com/documentation/en-us/red_hat_enterprise_linux/9/html/9.1_release_notes/technology_previews KTLS available as a Technology Preview RHEL provides Kernel Transport Layer Security (KTLS)..
Ive got a handful of servers running asterisk 16 currently with voicemail stored in a database via ODBC.Some users register to multiple servers and Im having an issue with MWI.Specifically Polycom phones seem to not be able to use two different MWIsourc..
Hi. Asterisk 16.2.1 I have a dialplan where one context (named inbound) performs: Originate(Local/${Target}@inOrig,exten,inbound,${EXTEN},208) The idea is that this command will spawn a call to the context inOrig on the same machine, and then ret..
am using asterisk 18.14.0 with pulse audio and dialing console dsp and getting a warble or a clipping in my audio.This is my cli log== Using SIP RTP CoS mark 5 > 0x7f47b80132a0 — Strict RTP learning after remote address set to:192.168.1.8:19436– Execut..