On 8/20/10 7:24 PM, Martin Steinmann wrote:
Joegen
I think a possible solution close to what you are suggesting is
captured here. This would likely be the most elegant solution.
http://track.sipfoundry.org/browse/XX-5010
Solving the problem described here might be easier and get us part of
the way. Eventually we will need to address both.
http://track.sipfoundry.org/browse/XX-4818
http://track.sipfoundry.org/browse/XX-8692
to further complicate matters are pure sip url calls from 'the world'.
If you are forwarding 5060 -> sipxbridge:5080 and 5061
->sipxbridge:5060, what happens when I make a sip:[email protected] to
you? do I have to use sip:[email protected]:5061?
I assume all this goes away if we put a REAL sbc in front of sipx?
anyone know if the freeswitch package for pfsense can be used to do
this? or sipproxyd? this is what you are talking about.
so, bottom line, if the ITSP wont' signal on port 5080, its a no go for
now, right?
ps, remote users:
I am not convinced this works. I tried a cisco (you yelled at me) I
bought two polycom phones to test. still can't get remote to work right.
will document that later . polycom 335. 3.2.1 firmware (no choice) you
put a call on hold, you can't pick it back up (no audio).
you transfer a call (which puts it on hold), you lose audio.
firewall problem? yeh, right. blame the firewall now.
the cisco remote didn't have those problems. all the cisco did was lose
audio after 5 mins. must be the firewall.
--
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