Re: [asterisk-users] Asterisk question
2013/2/20 Nguyễn Công nguyencong.1...@gmail.com Hello everyone, I’m new to Asterisk and I have a question. There is a phone call between two users, then they are talking to each other directly or by the server. I mean all packets from the user A to user B will be send directly to each other or will those packets from user A must be send to server and server will send to user B. Thanks. -- Both cases can happens. In a VoIP call we have two connections, one is used for signaling, usually port 5060 for SIP protocol, UDP transport and one is used for media (voice), usually random port. When the call starts the asterisk server sits in the middle of the media path, meaning all voice packets from phone A go to asterisk server and they are rerouted to phone B. After few milliseconds, if configured this way, asterisk server instructs the phone A to send the media directly to phone B to save bandwidth. It is named reinvite Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk question
Hello everyone, I’m new to Asterisk and I have a question. There is a phone call between two users, then they are talking to each other directly or by the server. I mean all packets from the user A to user B will be send directly to each other or will those packets from user A must be send to server and server will send to user B. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk question - media handling
On Wed, 20 Feb 2013, Nguyễn Công wrote: There is a phone call between two users, then they are talking to each other directly or by the server. I mean all packets from the user A to user B will be send directly to each other or will those packets from user A must be send to server and server will send to user B. Depending on the technology (IAX or SIP) and the configuration, you can choose to have the Asterisk server handle the media (RTP) or not. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk question
Any idea how to read an external file, grab some stuff and push it back into an Asterisk variable? I can do it the other way with: system(echo ${UNIQUEID} = /home/ast/curr_calls) but I'm a bit stumped on how to go the other way around much thanks, Paul Hales ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users