2016-03-04 18:59 GMT+01:00 Richard Mudgett :So, if the patch gets committed to trunk, then using the following should do it :CALLERID(pres)=unavailable CALLERID(pres)=prohibThanks Richard for this prompt and valua..
Author : Olivier
Home » Articles posted by Olivier (paged 14)
Ive read SIP Connect 2.0 draft lately.It mentions specific use if either of the following values is present in the From: field of an INVITE message. The values are:sip:unavailable@unkown.invalid sip:anonymous@anonymous.invalidIm using Asterisk 13 ..
Thank you all for pointing me in the right direction.Now I learned I have to care about MTU.Best regards2016-03-03 21:27 GMT+01:00 Toufic Khreish..
Im remotely managing an asterisk setup using an OpenVPN client on this Asterisk box, connecting to an OpenVPN server of mine).This box is mainly connected to PSTN. It is also connected to the Internet, only for remote management.The former ADSL l..
Im currently evaluating if it would possible/not too difficult to build and maintain an automatic attendant application.More precisely, my requirements are:- must work with Asterisk- should be installable on debian or CentOS- works this way :. cal..
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..
Im discovering Digium D70 phone (1.4.2 firmware) with asterisk 13.When pressing phones Msgs button, I see its dialing extension 800 no matter what I edited in my phone config file. Relevant piece of phone config is:Ive tried in the above voicemail fie..
Ive recently given a try to a Digium D70 phone.At the moment, Im configuring them though config files with a DHCP server and not using DPMA. Of course, Im connecting them to Asteris (PJSIP stack on 13.7.0).Which is the best place to:- read about p..
Ive got a request from a prospective customer demanding a SIP hardphone able to provide 120 BLF to its operator.Each BLF should show current extension activity (blink when ringing, …)and allow speed dialing.Beside finding hardware matching these requiremen..