<div dir="ltr">In order to have a second sip profile for the DIDs that means you would have to specify a different port for each profile, as far as I know.<div>Is there a one-port solution for incoming from DIDs and registered users?</div>
<div>-Avi<br><br><div class="gmail_quote">On Fri, Nov 19, 2010 at 11:53 AM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#FFFFFF"><div class="im"><div>"<span>Restart the profile and try again. NOTE: this leaves you pretty much wide open, so I would recommend you make sure that your dialplan is locked down so that you don't get toll frauded..."</span></div>
<div><span><br></span></div></div><div><span>Neil, a good way to avoid this would be to have two sip profiles - one for DIDs and one for registered users. Give each it's own dialplan context. That way the DID dialplan context can only allow the DIDs, while the registered user context can be a bit more flexible.<br>
</span><br>Steve on iPhone</div><div><div></div><div class="h5"><div><br>On 18 Nov 2010, at 23:00, Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>> wrote:<br><br></div><div>
</div><blockquote type="cite"><div>Are you having them come into the internal profile? (i.e. port 5060?) If so then make some changes:<div><br></div><div>In conf/sip-profiles/internal.xml remove line: </div><div><param name="apply-inbound-acl" value="domains"/></div>
<div><br></div><div>In conf/vars.xml change this line to false:</div><div><X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/></div><div><br></div><div>Restart the profile and try again. NOTE: this leaves you pretty much wide open, so I would recommend you make sure that your dialplan is locked down so that you don't get toll frauded...</div>
<div><br></div><div>-MC</div><div><div><br><div class="gmail_quote">On Wed, Nov 17, 2010 at 2:26 PM, Neil J. Macvicar <span dir="ltr"><<a href="mailto:neilm@kernelspace.co.uk" target="_blank"></a><a href="mailto:neilm@kernelspace.co.uk" target="_blank">neilm@kernelspace.co.uk</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<br>
<br>
I have recently compiled and installed freeswitch, and it is running with no problems. I can make and receive calls via my defined gateway with my provider, VoipTalk.<br>
<br>
My problem lies in that I have several DIDs with Voiptalk, and I have the option of routing them directly to my IP address as a SIP call. When I do this and call one<br>
of the numbers, not much happens. So I put sofia into debug sip mode, and observed the same message repeating :-<br>
<br>
------------------------------------------------------------------------<br>
send 866 bytes to udp/[217.14.138.126]:5060 at 22:22:15.236893:<br>
------------------------------------------------------------------------<br>
SIP/2.0 407 Proxy Authentication Required<br>
<br>
217.14.138.126 being one of my providers' machines.<br>
<br>
What i'd like to know is how can I get freeswitch to stop asking for authentication to my provider, without putting a blacket "allow" in the domain ACLS.<br>
Voiptalk have a large number of IP addresses that they use to send the outbound calls, so listing them all would be a chore.<br>
<br>
Thanks in advance,<br>
--Neil.<br>
<br>
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