<proxy>

USECALLMANAGER.nz

</proxy>

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

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.
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)d(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,om(224.0.1.1)d(From ${CALLERID(number)}))