[Freeswitch-users] Newbie question

[David Ponzone]

Michael,

with FS, you do what you want.
The default conf provided uses 5080 for external profile, 5060 for  
internal, but you can totally change that (I do), and anyway, what you  
do of one profile is totally up to you.
External/internal are just names you can also change, they don't not  
imply anything.

David Ponzone  Direction Technique
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Le 23/08/2010 à 17:56, Michael Scheidell a écrit :

> 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>
>
> 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
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> d: 561-948-2259
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> sip:michael.scheidell at secnap.com
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