Archives : February-2016
, 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..
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everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). Were trying to upgrade to 11.13.1 (The Debian Sta..
one of my client have hundreds of siemens openstage phones i want implement provisioning (1) for Asterisk and donate the code to some OSS provisioning projectcan you recommend some live provisioning project?thanks(1) http://wiki.unify.com/images/c/c7/OpenStage_Provisioning_Interface_Developer%27s..
all, Can someone recommend what hardware to use for a 1000 analogue line capacity asterisk PA..
all, I am currently using asterisk 11, and I am trying to figure out how to set the uri parameter telephone-context. I need to set it for outbound calls for a specific carrier when making emergency calls and dont seem able to find the option to set it.Rega..