Archives : February-2017
Asterisk 13.3.2I change the allowed codec from ulaw to g729 in sip.conf and enter sip reload on the console, but calls continue to use ulaw until restart.Before reload:lc10*CLI> sip show settings Global Signalling Settings:—————..
Asterisk 13.3.2The console command sip show settings shows the allowed codecs in the Global Signalling Settings but does not include the packetization setting.Similarly, both core show channel and sip show channel will show the codec(s), but not ..
all,we have a running Asterisk 11.25.1 in a VM (qemu/kvm) OS being Debian 7.11 (wheezy), the host OS being the same.Problem: when we restart the server (eg host + VM), all customers Asterisk connecting without a VPN (doesnt matter which Asterisk versi..
Got a strange situation[ext-queues]… exten => h,2,ExecIf($[${CALLERID(num)} = ]?Set(var29=${SHELL(curl -XPOST –header Content-Type: application/json –header Accept:application/json -d {\Phone\: ${FROMEXTEN}, \Source\: \asterisk\} http://sIte.com:80/api/v1/calls?apiKey=UABVAEI&clientId=3)}))ex..
Theory: The carrier is not responding with 100 Trying in the expected time. Hence, Asterisk is sending the INV..
I have two upstream providers we use for US termination.The dialplan sends calls out the primary and if that fails for specific reasons, it sends the same call out the secondary. This has worked well for us when we are lazy about keeping balances ..
Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS?Could anyone provide pros/cons for the various ASR options for Asterisk?We need the ability for very large grammars (over 100,000 options).Because of this, my initial thou..
All;I am running Asterisk 11.6-cert16 and I have voicemail setup so voicemail messages are sent as email attachments. That works fine. However, the body of the email contains the CallerID(name), but is missing the CallerID(num). For example, the em..
I have some troublesome numbers that I would like to capture the SIPdialogue when I am calling them. When I am about to dial the number, is there any way to turn on SIP debugging in the dial plan before I make the call? (and turn it off after the c..
Years ago, I used Monit to monitor Asterisk and restart it whenever it failed.Now, I wonder which tool I should pick for an Debian 8 (current) or CentOS7 (future) environment. The main reason Im looking for this tool is to avoid as much as possib..