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.<br>
<br>-Steve<br><br><br><br><div class="gmail_quote">On 23 August 2010 16:56, Michael Scheidell <span dir="ltr"><<a href="mailto:michael.scheidell@secnap.com">michael.scheidell@secnap.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div text="#000000" bgcolor="#ffffff">
looked at FAQ's first, just want to clarify.<br>
Some background, looking at replacing a running sipx install due to
limitations on ITSP's<br>
(conflicting ITSP and sipx limitations)<br>
Biggest limitation I see is that sipx without an external SBC forces
the ITSP to send calls to port 5080.<br>
(Big Sip trunk providers like ATT, Level3 and Verizon don't want to do
that. I actually understand why)<br>
<br>
in the firewall ports section it notes port 5060 for 'default'
internal, 5070 for default nat and 5080 for default external.<br>
<br>
I suppose the biggest question is can I set up freeswitch to listen for
SIP inbound trunk calls on port 5060 without loosing functionality<br>
(with sipx, if I do this, I can't forward calls:<br>
<a href="http://list.sipfoundry.org/archive/sipx-users/msg23554.html" target="_blank"><http://list.sipfoundry.org/archive/sipx-users/msg23554.html></a><br>
<br>
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)<br>
<br>
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)<br>
<br>
POC would be to use someone like <a href="http://voip.ms" target="_blank">voip.ms</a> and static registration (which
uses port 5060). and have full regression testing. inbound calls,
outbound calls, calls transfterd, (correct caller id shows up!)<br>
<br>
Currently, this only works on <a href="http://voip.ms" target="_blank">voip.ms</a> user/password authentication
since the invite/auth exchange will register port 5080.<br>
<a href="http://voip.ms" target="_blank">voip.ms</a> 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.<br>
<br>
(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.<br>
on sipx, I added a two digit 'trunk' code '**' to make outbound ISN
calls)<br>
<br>
<div>-- <br>
Michael Scheidell, CTO<br>
o: 561-999-5000<br>
d: 561-948-2259<br>
ISN: 1259*1300<a><br>
sip:michael.scheidell@secnap.com</a><br>
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