Add rtppage() application to do multicast or unicast RTP paging to SIP phones.
[asterisk/asterisk.git] / configs / rtppage.conf.sample
1 ; Configuration for the rtppage() application 
2 ; that sends audio in multicast or unicast mode to phones
3 ; for paging
4
5 [general]
6 ttl=10
7 tos=ef
8
9 [testgroup]
10 type=basic
11 rtp_address=192.168.83.147
12 rtp_port=12346
13
14 [linksysgroup]
15 type=linksys
16 rtp_address=224.168.168.168
17 rtp_port=34567
18 control_address=224.168.168.168
19 control_port=6061
20