Hi! > It looks like RTP has a real problem with timing if it is not receiving > RTP packets. If the outside call that is placed on hold is not generating > any audio, the sip/fxo gateway does not send * RTP packets. > Is this valid?
Yep, unfortunately. That's why for example in X-Lite you'll need to change settings to "Transmit Silence=Yes". No clue how to do that on the GS, I don't own any of these. Philipp _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
