Archives : February-2017
I operate an Asterisk server (v11.13.1) on Debian stable, and its rock-solid. The other day, however, I accidentally upgraded the kernel from the stable 3.16.0 to 4.9.0. Subsequently, audio stopped working.Below you can find my analysis while runn..
my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My extensions.conf file was mostly copied from server running Asterisk 1.8. That being said! If I dial a number and get a busy signal I get the following error:– SIP/voipeer-00000..
if a PJSIP call fails, how can I capture SIP code, like 503,603 etc?in old SIP channel, we had ${HASH(SIP_CAUSE,)}but in PJSIP it has to be the outbound channel, which is gone when the control returns to the callin..
i have similar problem to https://issues.asterisk.org/jira/browse/ASTERISK-25806do you know about some workarounds/patches for better scalability?t..
I need to make calls to a list of numbers one at a time and once the user pick the phone connects to an IVR where I can get few data, aftera call finishes the 2nd number get called and so forth.Im familiar with Asterisk/Elastix but the Campaign feat..