There's an option for the Polycom phones to switch the hold behaviour.
Set voIpProt.SIP.useRFC2543hold to 0, and it should use RFC3264 rules for signalling hold instead of 0.0.0.0.
Benko wrote:
Hello!
I'm having a issue with NAT and rtpproxy. Usually my setup works fine with natted clients, the Connection Information is overwritten with the IP of the rtpproxy and audio passes through in both directions. However, today i came across a problem where the Polycom 501 sets a outgoing ip of 0.0.0.0 instead of the private ip after resuming a call that was on hold(actually, the other party is invited again) - and the force_rtp_proxy ()-command on openser left the ip untouched instead of overwriting it with the rtpproxy-ip. As a result the person that was on hold had audio but the polycom user(with the "wrong" ip) hadn't.
The false ip left aside, is it expected behaviour of force_rtp_proxy to not touch 0.0.0.0?
Just out of curiosity - does someone know the "on hold"-problem with polycoms?
thx christian
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