Archives : October-2018
Im curently setting a lab environment for load testing an Asterisk instance.This environment includes:- a management workstation where I would like to run scripts andstore test reports- a box hosting SIPp- the Asterisk box Im load testing (System Un..
I just noticed this upon startup since updating from 15.6.1 to 16.0.0 – do any of these matter? [Oct 18 12:12:18] WARNING[4489]: loader.c:2228 load_modules: Some non-required modules failed to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modul..
Lets say I have a conference room of 8 users. At some point in the evening, we need to hook up with a Zoom conference. That means hooking up that existing pool of users to a new PJSIP channel. An admin would dial in, enter a pin, and initiate that connecti..
   It seems that app_swift does not work with Asterisk 15 or 16. I just get errors when trying to compile: [root@pbxoficina app_swift]# ./configure checking gcc… checking swift… checking asterisk… creating Makefile  *****************************************************..
Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it should. Thank you,IvanMessage: 2Date: Mon, 15 Oct 2018 23:39..
You can ask your provider to accept PAI headers that you Would add to your SIP Invite request.Usually, this is what you do when you want to block Your caller id from showing it to the callee. The only way that the provider can identify you (for bill..
I have an Asterisk system with 2 trunks (as shown below).I need to be able to disable a trunk at runtime. I may not change the dialplan but I can change sip.conf and reload. Any attempt to dial in the dialplan uses trunk A and trunk B in that ord..
Hello! Just upgraded asterisk from 13 to 16 and found that php-agi library is not compatible. It waits for –END COMMAND– after command is completed, but, as I see from tcpdump, now asterisk does not send such string after command is completed. Co..
, I have a queue in which I add a member located outside the company and connected to an outside asterisk. Lets say peername is ABCD123. In the queue I gave SIP/ABCD123 as interface which is not existing on the local asterisk. Is there a way to conn..
We have following problem. On some of the extentions I call cell phone after 10 seconds or so.Or, like this one below- we call cell and office phone at the same time;Eric on extension 105exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)Â Â Â Â Â Â..