Category : Asterisk Users
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,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..
I am running ubuntu 20.04 fully patched along with Asterisk 18.8.0This is a VM environment with VMWare.I found this in the logs today.[1768362.083207] CPU: 2 PID: 1939739 Comm: asterisk Tainted: GOEL5.15.0-52-generic #58~20.04.1-Ubuntu[1768362.083209]call_cpuidle+0x23/0x50[1768362.083217]do_idle+0x1f4/0x270[1768362.0832..
Trivial issue. I have a script to rebuild asterisk with the following line: menuselect/menuselect –disable MENUSELECT_MOH –disable CORE-SOUNDS-EN-GSM –enable CORE-SOUNDS-EN-WAV –enable app_macro –enable codec_opus –enable chan_phone –enable chan_..
We have a problem where Asterisk is resetting the CSeq on a re-INVITE, and the phone receiving the re-INVITE is rejecting it, probably as a result of that. Would anyone be able to offer any insight please?The scenario is:Phone A makes call 1 to Aster..