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...his manner. The only issues I get are those of bandwidth availability or rather occasional lack of it. Hosted PBX's are no different. The hosting service should be providing a similar mechanism (although it might not be Asterisk based). Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Michakl Gaudette wrote: > Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk > server somewhere where there was no NAT for the * box that the SIP phones > wouldn't create any issues. > > How do you people with Hosted PBX handle the deployment of SIP phones beh...

...ust elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect the ports of the router as well. Or you can configure your SIP phone to use a STUN server. Please read in voip-info.org about SIP NAT, there are good suggestions. regards On 1/20/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote: > Hello, > > I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my > wholesale provider. That worked, fine. I ahd to open up the ports on my > router, forward them to the correct box, again fine. > >...

That was exactly it! Thanks you VERY much! Mike ---- For the sip setting in sip.conf that setsup your voip provider add: canreinvite=no On 2/6/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote: > > Hi, > > I've had a bit of a problem with one way audio, and it happens exactly when > I believe it shouldn't (and works perfectly when I would guess I could have > issues. > > Setup: > GrandStream GXP2000---...