[isapros] Re: Publishing Phones
- From: "Thor (Hammer of God)" <thor@xxxxxxxxxxxxxxx>
- To: <isapros@xxxxxxxxxxxxx>
- Date: Fri, 24 Aug 2007 14:49:28 -0700
Except that RTP won't work with NAT unless it's symmetric RTP (I think).
t
> -----Original Message-----
> From: isapros-bounce@xxxxxxxxxxxxx [mailto:isapros-
> bounce@xxxxxxxxxxxxx] On Behalf Of Jim Harrison
> Sent: Friday, August 24, 2007 1:02 PM
> To: isapros@xxxxxxxxxxxxx
> Subject: [isapros] Re: Publishing Phones
>
> Can the internal device operate as a SecureNET client to ISA?
> If so, an access rule allowing RTP should work.
> In order to allow inbound connections, you'll have to server-publish
> the
> internal phone.
>
> -----Original Message-----
> From: isapros-bounce@xxxxxxxxxxxxx [mailto:isapros-
> bounce@xxxxxxxxxxxxx]
> On Behalf Of Amy Babinchak
> Sent: Friday, August 24, 2007 11:50 AM
> To: isapros@xxxxxxxxxxxxx
> Subject: [isapros] Re: Publishing Phones
>
> SIP doesn't seem to be the problem in this scenario. It's RTP outgoing
> that seems to be the problem. The Trix box has its own SIP proxy and
> that end of things works.
>
> -----Original Message-----
> From: isapros-bounce@xxxxxxxxxxxxx [mailto:isapros-
> bounce@xxxxxxxxxxxxx]
> On Behalf Of Jim Harrison
> Sent: Friday, August 24, 2007 11:24 AM
> To: isapros@xxxxxxxxxxxxx
> Subject: [isapros] Re: Publishing Phones
>
> " SIP phones ".
> ISA don't "do" SIP (yet).
> You might want to try CollectiveSoiftware's LCS Filter.
> They say it's only for LCS scenarios, but because it's "SIP-aware", it
> may well help (and you can try it for free).
>
> -----Original Message-----
> From: isapros-bounce@xxxxxxxxxxxxx [mailto:isapros-
> bounce@xxxxxxxxxxxxx]
> On Behalf Of Amy Babinchak
> Sent: Friday, August 24, 2007 7:39 AM
> To: isapros@xxxxxxxxxxxxx
> Subject: [isapros] Publishing Phones
>
> I'm trying to help out a guy with an Trix box. I haven't worked with
> any
> VOIP systems yet. Anyone run into this situation? He says it's the
> outgoing end that's the problem so it seems like that should be easy
> enough to fix.
>
> "The scenario is two SIP phones, one internal inside the firewall and
> the other external outside the firewall. The external device must be
> able to register with the SIP Server through the firewall, then the
two
> devices must be able to send audio directly to each other.
>
> The SIP session is negotiated via the SIP Proxy (i.e. the Asterisk
> virtual PBX), that part seems to work OK - I can publish my SIP server
> through ISA and that works as expected. Then the telephone devices are
> supposed to communicate directly using RTP. This is where it goes
> wrong.
> Oddly enough though, it is always the OUTGOING part of the
conversation
> that I can't make work. I would expect it to be the INCOMING part that
> didn't work, but that always works just fine."
>
>
>
>
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