I’m running Asterisk 184.108.40.206 with a couple of softphones registered.
We’re using direct RTP flow from endphones when the call is active.
So far, when a softphone pauses the call, Asterisk doesn’t really pause the
end phone but just updates the stream changing the media address from the
softphone that asked for pause to his own address.
The SIP specifications tell that a SIP hold is characterized from a
SIP INVITE with either the 0.0.0.0 as media address and/or a sendonly in media attribute.
The problem is that the softphone sends those fields properly but Asterisk changes
them to his IP and sendrecv before sending the invite to the end phone.
Is there a way to have the SIP HOLD be sent to end phone?