RTP Streaming
Audio can be directly streamed one-way to a Cisco phone using the newSIPCiscoPage() application either via a unicast or multicast RTP stream.Options
The following application options are available.| m(address) | Stream audio to the specified multicast IP address. |
| p(port) | Port on the phone to receive the audio, an even number between 20480 (default) and 32768. |
| v(volume) | Set the receive volume percentage (1-100) on the phone's speaker. |
| d(message) | Display a message in the status line of the phone for 10 seconds. |
| a | Play an alert beep on the phone. |
| o | Indicate that peers are off-hook when paging. |
| b | Page the phone even if it is busy (Off-Hook, Ringing, Busy or DND). |
Unicast Audio
Unicast uses a separate RTP audio stream for each phone.; Unicast page 301 and 302 and set speaker volume to 75%
exten => 380,1,SIPCiscoPage(301&302,ov(75)ad(From ${CALLERID(number)}))
Multicast Audio
Multicast uses only a single RTP audio stream for all phones. You must specify a multicast IP address.; Multicast page 301, 302 and 303
exten => 380,1,SIPCiscoPage(301&302&303,oam(224.0.1.1))
External Streaming
Themediastream script from the commands archive below can be used to stream a .wav file to multiple phones. See CGI Execute for more information.file_download commands-1.2.tar.gz (13K) event 05/11/2020 security SHA256:dd1292914956fe544e26dd4cd60a7f57dde405a36b10ccd364e0cb8028005b4b.