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Thank you for the info now it works, but I have one other problem.<br>
<br>
When I call from voip phone to the outside network the call works just
fine, but when I call from outside network to the voip phone (connected
to freeswitch) I can answer the phone, but there is no rtp between, on
neither side. <br>
<br>
Wireshark and Cisco debug ccsip shows that freeswitch returns <a class="moz-txt-link-freetext" href="SIP:415">SIP:415</a>
Unsupported media type<br>
<br>
Anynone having any clues?<br>
<br>
I can include lot more debugs if needed?<br>
<br>
<br>
Brian West wrote:
<blockquote cite="mid:280F5B85-2D62-46F3-A07D-51EAF3ED1EF0@mac.com"
type="cite">You want to send calls to port 5080. Thats the bottom
line since you have phones registering you want to keep that.
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<div>sip_profiles/default.xml (for phones to register with NOT
gateways to send calls to, runs on port 5060)</div>
<div>sip_profiles/outbound.xml (all in/out interaction with
registered gateways and anonymous inbound sip, runs on 5080)</div>
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</div>
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</div>
<div>outbound.xml is setup to hit the public context
(dialplan/public.xml)</div>
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</div>
<div>Their are examples in there on how to route a did.</div>
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</div>
<div>/b</div>
<div>PS: The Default config are NOT ment to be the only way to
configure and run your system. Its a general config setup to demo how
to setup FreeSWITCH. </div>
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<div>On Mar 10, 2008, at 9:37 AM, Josip Djuricic wrote:</div>
<br class="Apple-interchange-newline">
<blockquote type="cite">
<div bgcolor="#ffffff" text="#000000"> Thank you Brian,<br>
<br>
would it be possible maybe to go little more deeper explaining more in
depth these two profiles?<br>
<br>
Brian West wrote:
<blockquote cite="mid:E3CF5E96-D7CA-4562-8772-7BD27F932DA5@mac.com"
type="cite">
<pre wrap="">You need to understand why and how this works:
sip_profiles/outbound.xml vs sip_profiles/default.xml
If you have phones registering then you're better off sending calls to
port 5080 into the public context as that profile doesn't auth.
If you really wanna do this then open up default.xml sip profile then
set auth-calls=false then change the context= to default.
That should get you going but understand that opens you wide up.
/b
On Mar 10, 2008, at 7:48 AM, Josip Djuricic wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hello,
I'm testing freeswitch with AS5300 for inbound and outbound call's.
Does anyone have any info on how to disable Proxy authentication,
because when the AS5300 sends the call to freeswitch, debug on AS
said:
Received: SIP/2.0 407 Proxy Authentication Required.
I tried to turn off gateway authorization, and setting authless
authorization, but it just doesn't work.
Thanks
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