Hellousing Asterisk 18.104.22.168.What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ?So I can log this information.Kind ..
Asterisk Project Security Advisory – AST-2017-003 ProductAsterisk SummaryCrash in PJSIP multi-part body parser Nature of AdvisoryRemote CrashSusceptibilityRemote Unauthenticated Sessions Severity CriticalExploits KnownNo Reported On13 April, 2017 Repor..
Asterisk Project Security Advisory – AST-2017-002 ProductAsterisk SummaryBuffer Overrun in PJSIP transaction layer Nature of AdvisoryBuffer Overrun/CrashSusceptibilityRemote Unauthenticated Sessions Severity CriticalExploits KnownNo Reported On12 Apr..
We use Snom870s together with Asterisk 13.14.0 and FreePBX22.214.171.124.I am having an ongoing problem with setting the BLFvalues on these phones from the configuration file generated from FreePBX.In FreePBX we employ the Commercial Endpoint Manager (C..
i have strange problem with asterisk 13.15.0+pjsip bundled/CentOS 7/systemd start scriptwe are using chan_pjsip only for webrtc endpoints . switched from sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days agotoday i have problems with stopping/crash..
all,Its slightly OT, but hopefully someone can help. Im struggling with getting Cisco IP Phone 7942G to fail over to our secondary Asterisk server in the event of a primary failure.We recently bought a bunch of new Cisco 7942G phones, which now c..
I need to have an extension on a SwitchVox server dial out to one on an Asterisk (FreePBX actually) box which will host a voice directory. The Asterisk server will then need to dial one of the SwitchVox extensions if it gets a voice match.Anyone ..
We run Asterisk 13 using the FreePBX 126.96.36.199 distro based on CentOS-6.4.We also run HylFAX+ 5.5.3 with iaxModem 1.2.0 on the same system with AdvantFAX as the web front-end.Our two fax lines are configured as iax2 DEVICES.These components have b..
Im trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures.For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesntS..
This is your friendly 6 month warning that Asterisk 11 will be reaching an official end of life state on October 25, 2017.As many of you know, for the past 6 months Asterisk 11 has been in security fix only mode.This means it currently does not rece..