Archives : February-2016
I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio gateway. I am able to make calls outbound through the gateway, but I am not able to make calls into the PBX from external PSTN.Specifically, an incoming call is _received_ by Asteri..
Tonight community services may have intermittent availability due to maintenance. This maintenance will begin at approximately 8:00 PMCST[1] and should last no longer than three(3) hours, ending around11:00 PM CST.The affected services are:* All Aster..
All, Ive been wondering if I can instruct asterisk in the dialplan to engage the Media handling for a particular call or not.Ive SIP users behind Kamailio & RTPProxy, and I can make use of sip.conf setting directmediadeny|directmediapermit to offl..
, Ive been using Grandstream phones for more than 10 years, but only yesterday tried to use Early Dial… and I failed. What is needed on the Asterisk side to reply 484 to INVITE? Phones are talking to chan_pjsip on Asterisk-13.7.1. Thanks, – — Jean-De..
HiWe are using asterisk 1.8.23.1 on CentOS 6Is there a way that transferring by SIP REFER can be blocked on a call by call basis?..
Im having my first steps with WebRTC.Ive found this line in http.conf.sample (asterisk 13.7.0):;tlsprivatekey=; path to private key file (*.pem)only.Is it a typo ?I expected something like:;tlsprivatekey=; path to private key file (*.key)on..
Im trying to have my first calls with WebRTC. My server has asterisk 13.7.0.Im following the instructions from the wiki [1]. So Im using [2] live demo from a Chrome navigator (v48) on Debian Jessie station.Whenever I type something like ws://123.123.123.123:8088..
..