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 link
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. |
| t(seconds) | Number of seconds to display status line message, default is 10. |
| 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 link
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 link
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 link
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-2.8.tar.gz (17K) event 01/11/2025 security SHA256:8afd0369aaacc9338180b2efbd607d7b103e9cb331ca8314b540748c30721f77.