Archives : April-2020
In the context of the COVID-19 pandemic, Nasim Telecom decided to publish the first six-captures of Persian Asterisk book that was written by me and Mr Najafi, published in 2017. Also it was introduced by Mr. David Duffett in AstriCon 2017 on Flori..
how do I make a bug report?I filled in the form to make a report and https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues reported by me. If someone knows how to get asterisk to re-register when using pjsip after the registrat..
Gang Not a specific Asterisk Question. But I wonder, if the called party replies with 183 + SDP indicating support for telephony-event. Should the caller be able to send DTFM Tones? Swiss Railways uses an IVR that kicks in before the call is answer..
We have a provider which is using Kamailio as front end. Our asterisk 13/chan_sip server has no problem to register and pass/receive calls form this provider. Now we want to move to asterisk 16/pjsip and face problem. Registration is OK but when we p..
I am running Asterisk 16.9 on FreeBSD 12.1-RELEASE-p1.I keep seeing lines like this in my logs. [Apr1 13:30:33] NOTICE[101155][C-00004526] chan_sip.c: Call from (45.143.220.235:5356) to extension 2037 rejected because extension not found in context unauthenticat..
All,Does anyone have stats on how many callers they can get on a single ConfBridge per core? I was testing on Digital Ocean with 30 users in a room and I was shocked by how well it kept up with 40 callers. I am trying to figure out how scalable it..
Im compiling an Asterisk system on a ESXi VM with recent CPU, but will deploy onto an old ESXi VM with older CPU. Is it possible to configure Asterisk to NOT use CPU specific instructions/optimizations so that the executable is portable? ThanksDan..
everyone, I have a Atcom AX-1600P(1) card with a FXO module and I cant configure it. I have four extension with this PJSIP settings: — /etc/asterisk/pjsip.conf — [transport-udp] type=transport protocol=udp bind=0.0.0.0 [6001] type=endpoint transport=transport-..
– Is this list still alive ?How do I get audio/video into gstreamer from an asterisk call?T..
Le 26/03/2020 à 21:50, Kai Herlemann a écrit : Kai Hangup is h extension. your macro will never be executed. Solution: same = n,Dial(whatever) same = n,[…]) same = n,Hangup exten = h,1,1,DumpChan() same = n,System(/home/asterisk/bash_test)..