RTP Streaming

Audio can be directly streamed one-way to a Cisco phone using the new SIPCiscoPage 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 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 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(

External Streaming link

The mediastream 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-2.5.tar.gz (15K) event 08/05/2024 security SHA256:524313469bdddd19304eba4a40457b6ad1fbdff58dd627d5aaae44e446e4004a.