Archives : November-2015
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Im not getting any ringing when I use option r with Dial:Dial(DAHDI/1-1, motif/8447/+1@voice.google.com,,rTt) in new stackOtherwise all works. The call goes through, good ..
I save to drive a fancy car by buying all the imitation luxuries from thi..
is there somebody using systemd start script on fedora/CentOS7 + asterisk 13 in production?i have strange problem with high cpu usage when asterisk is started via systemdthanks for feedbackp.s. systemd script is not in vanilla asterisk. only in fed..
allIm running a 1.8.32.3 instance of Asterisk on CentOS 7.About 80 SIP phones connected.Im getting tens of thousands of error messages in the CLI, and callers are having extreme difficulty dialing, they can dial once and then if they try to dial ag..
Guys I am seeing this error a lot in the CLI latelyWhat does it mean?Prodding channel SIP/..
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The Asterisk Development Team has announced the releases of:DAHDI-Linux-v2.11.0-rc1DAHDI-Tools-v2.11.0-rc1dahdi-linux-complete-2.11.0-rc1+2.11.0-rc1This release is available for immediate download at:http://downloads.asterisk.org/pub/telephony/dahdi-li..
SoI am using PJSIP realtime with Asterisk 13. I set the qualify_frequency column AORS and it now shows the RTT in milliseconds in the console. I want to be able to display that in a webpage, and was hoping the RTT would be updated in one of the realt..
I am searching for a solution to encrypt authentication from Asterisk server to clients. Searching srtp seem to encrypt traffic, I just want client authentication with encryption. Can someone point to the right direction? has anybody used ZRTP? experie..
All,I have configured WebRTC according to the install document.The clients register correctly. Im use SIPjs. The clients are able to send messages to the server. The SIP debug shows the messages being received. However Im stumped for directions on ..