Hmm, That's likely. Here's some info from voip.ms:
SIP: 5060 UDP RTP Range: 10001-20000 UDP
Is there a way to make voip.ms match sipx or do I have to change sipx to
match voip.ms (and change my finewall)?
When I set up voip.ms, I used this how-to from Tony. However, some
things have changed in sipx so there is not a one-to-one match.
http://blog.myitdepartment.net/?p=191
One of the things that is no longer in sipx (at least not that I could
not find - I'm using v4.2.1) is the ITSP template drop down which he
says to use. My guess is this is where the modifications to make sipx
and voip.ms match were done.
If I have to change sipx to use port 5060 instead of 5080, does that
change how I support remote phones?
Here is what currently works:
* From the AA, I can transfer the call to any extension.
* I can call out
* I can transfer a call from one ext to another
* I can setup a conference call
* I can leave voice-mail
Stiles
On 03/26/2012 12:30 PM, Gerald Drouillard wrote:
Sounds like the call did not come in on port 5080
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