Archives : August-2019
Home » 2019 (paged 13)
Ive got an Asterisk 11.13.1 system running on a Debian Jessie platform. This systems extensions.conf doesnt include any reference to PJSIP, yet(only using chan_sip at the moment).This morning, it failed with:Aug 26 09:07:33 foobar kernel: [6534231.7764..
HelloI see on the CLI :tst*CLI> core show hints    -= Registered Asterisk Dial Plan Hints =-                     50@blf         : SIP/testacc7 State:Idle           Watchers 3                 Â..
All,Im using asterisk 16.4.0 with h264 and opus quite well using linphone 4.1client on android and baresip on linux.Im exploring use of h265 for improved video quality/lower network bandwidth. I do not see pass through support on asterisk for h265/hv..
We are running load capacity tests using Amazon AWS configurations.For the tests, we are basically scaling up calls to a second Asterisk box.First box that is calling the second box plays music on hold for 60 seconds, then hangs up the call. My init..
Hello! I encountered an outage of asterisk which showed like that: – 2019-08-10 07:22:21 Asterisk start – 2019-08-15 19:39:33 WARNING taskprocessor.c: The pjsip/outreg/ispPJSIP-00000060 task processor queue reached 500 scheduled tasks. – 2019-08-15 19:39..
All,We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered.Is there any possibil..
dahdi built fine on 5.1.20, but on 5.2.7: …………. CC [M] /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o SHIPPED /home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_6..
Who is the male voice artist who recorded the en_GB sounds for Asterisk? Would be useful to know in case of the need to get additional matching sounds recorded. Cheers Tony — Tony Mountifield Work: tony@softins.co.uk – http://www.softins.co.uk Pl..
We have a customer where their VM running Asterisk appears to have crashed.Fortunately, we had some debugging enabled. The asterisk messages file has this… (in notepad+ the blank line in the middle is all [NUL][NUL] [NUL][NUL]….)[08/12 15:30:55.8..
We have a system where two calls are in a ConfBridge with recording.This is Asterisk 16.3.0Channel A seems to work perfectly.Wireshark is showing the RTP to/from working fine and having no jitter/lag issues.This call hears everything from channel B.Chan..