[Freeswitch-users] Newbie question

[Steven Ayre]

Profiles are bound to IP + port, not just port. You could have an IP for
internal and a different IP for external, and then use 5060 on both. That
can either be either 2xLAN, 2xWAN, or 1LAN+1WAN IPs, it doesn't really
matter. Which IP calls go out from just depends which profile you use in the
dialstring.

-Steve



On 23 August 2010 16:56, Michael Scheidell <michael.scheidell at secnap.com>wrote:

>  looked at FAQ's first, just want to clarify.
> Some background, looking at replacing a running sipx install due to
> limitations on ITSP's
> (conflicting ITSP and sipx limitations)
> Biggest limitation I see is that sipx without an external SBC forces the
> ITSP to send calls to port 5080.
> (Big Sip trunk providers like ATT, Level3 and Verizon don't want to do
> that.  I actually understand why)
>
> in the firewall ports section it notes port 5060 for 'default' internal,
> 5070 for default nat and 5080 for default external.
>
> I suppose the biggest question is can I set up freeswitch to listen for SIP
> inbound trunk calls on port 5060 without loosing functionality
> (with sipx, if I do this, I can't forward calls:
> <http://list.sipfoundry.org/archive/sipx-users/msg23554.html><http://list.sipfoundry.org/archive/sipx-users/msg23554.html>
>
> I don't want to interfere with normal sip: url inbound calls, so I would
> expect (in an 'askerisk like' setup) to be able to receive sip calls to
> internal extensions as well as SIP trunk calls to port 5060.  (and be able
> to put both on hold, take them off hold, play moh to callers, transfer,
> blind and attended, and join a 3 way conference call)
>
> Next, looking for someone who has moved from sipx 4.2.0 to
> freeswitch/fusionpbx to quote me on a conversion (assuming I can use the
> above)
>
> POC would be to use someone like voip.ms and static registration (which
> uses port 5060).  and have full regression testing.  inbound calls, outbound
> calls, calls transfterd, (correct caller id shows up!)
>
> Currently, this only works on voip.ms user/password authentication since
> the invite/auth exchange will register port 5080.
> voip.ms has no way to specify the calling port, and even if they did, this
> is just the POC.  Level3, Att and Verizon will ONLY send to port 5060.
>
> (ps, ISN dialing.  seems to be tied in to e.164 dialing, but I was not able
> to actually tell if that is the case.
> on sipx, I added a two digit 'trunk' code '**' to make outbound ISN calls)
>
> --
> Michael Scheidell, CTO
> o: 561-999-5000
> d: 561-948-2259
> ISN: 1259*1300
> sip:michael.scheidell at secnap.com
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>
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