All; Please I need a help in the ooh323. First of all, the only way to have h323 working in asterisk 1.8.3 or 1.8.4 is to use ooh323? There is no way to get the normal h323 channel that come with asterisk to work fine !! Now, let us see the ooh323 prob..
From http://www.voip-info.org/wiki/view/Asterisk+presence Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a f..
When is the next release planned for as very keen to get it into Production but require the call pic..
The way I play a sound file into a bridged call is to use chanspy w option. I do this with an application that does AMI c..
list i need to create a call files in order to do a click to call with asterisk1.4 i want to use sip 223 in order to call phone number i have created a file.call in var/spool/asterisk/tmp and i move it to var/spool/asterisk/outgoing but there is no c..
Users, I would like to know about the RTP audio streaming. I am taking the example as youtube, in youtube if bandwidth is less the application will buffer and will stream the video; likewise how to do with audio buffering and play the file using ..
Is there a SIP header I can set (for Snom and Yealink phones if thats relevant) or any other mechanism to tell a phone to ignore a particular call from its missed call list? I have bits of the dialplan that ring groups of phones eg: exten => 200,1,Dial(Sip/112&SIP/113&SIP/1..