Hello!
I've previously posted this question on another community forum, as I wasn't aware of the existence of this one, which is, presumably, more suitable for this question. I tried deleting the previous one, but could not find how.
Preface:
I'm sending an RTPRx XSI command to a Cisco IP-Phone (8851), in the following format:
<CiscoIPPhoneExecute><ExecuteItem URL="RTPRx:<Local-IP>:<Phone-Port>:100"/></CiscoIPPhoneExecute>
Alternatively, I could send a <startMedia> command XML to phone's CGI/Execute interface.
Both methods successfully start a custom Rx (RTP) stream on the target phone, even while it's active in a call, and the audio streamed to it is well-heard at device's end (and only by it).
The IP Phone Services Application Development Notes document (for CUCM r9.1) says (page #86):
"If the user initiates the Rx or MRx URI using an application, then the user likely is not concerned about having the audio mixed with the current call."
But in fact, while the device is during a phone-call (regular, peer-to-peer), the audio streamed to the custom Rx stream is only heard at device's end, but not to the other call-party.
Question:
Is there a way to (deliberately) accomplish what's written in the doc., i.e. having the custom Rx audio mixed with current call, so that the other call-party hears it as well?
(possibly, using phone's BiB)
Possibly, the answer lies in the quote brought above - "If the user initiates the Rx or MRx URI using an application".
What does the phrase "using an application" mean?
Thanks much.
