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<div style="direction: ltr;font-family: Tahoma;color: #000000;font-size: 10pt;">Thanks, much appreciated.<br>
<br>
I'm seeing the Proxy-Authorization from Asterisk in tcpdump. It seems like I've been working against what's already built into Kamailio.<br>
<br>
Probably need to tweak some uri's though.<br>
<br>
When dialing out, the r-uri is:<br>
sip:mobilenumberhere@exampleip<br>
<br>
But the uri part of the Proxy-Authorization in the new INVITE ends up with uri="sip:mobilenumberhere@exampleip"<br>
<br>
However, I think it should be showing uri="sip:providerusernamehere@exampleip"<br>
<br>
Regards,<br>
<br>
Darren<br>
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<div style="direction: ltr;" id="divRpF31264"><font color="#000000" face="Tahoma" size="2"><b>From:</b> sr-users [sr-users-bounces@lists.sip-router.org] on behalf of Daniel-Constantin Mierla [miconda@gmail.com]<br>
<b>Sent:</b> Monday, 11 May 2015 6:07 PM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List<br>
<b>Subject:</b> Re: [SR-Users] Handling 407 Proxy Authentication, Elastix MT<br>
</font><br>
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<div>Hello,<br>
<br>
<div class="moz-cite-prefix">On 11/05/15 08:41, Darren Campbell (Primar) wrote:<br>
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<div style="direction:ltr; font-family:Tahoma; color:#000000; font-size:10pt">Hi all<br>
<br>
Have Asterisk listening on 127.0.0.1 and aiming to route all inbound/outbound SIP via Kamailio listening on 127.0.0.1 and external interface.<br>
<br>
Inbound calls from the SIP PROVIDER work just fine. Have NAT, rtpproxy configured for successful registration and subsequent INVITEs etc.<br>
<br>
Experiencing some challenges with the outgoing INVITES, primarily authenticating the outbound INVITEs.<br>
<br>
The current situation is this:<br>
Asterisk > INVITE > Kamailio > INVITE > SIP PROVIDER<br>
SIP PROVIDER > 407 Proxy Authenticate > Kamailio > Transaction Cancelled.<br>
Asterisk then plays number unavailable message.<br>
<br>
<br>
The desired situation is more like this:<br>
Asterisk > INVITE > Kamailio > INVITE > SIP PROVIDER<br>
SIP PROVIDER > 407 Proxy Authenticate > Kamailio > Asterisk<br>
Asterisk > INVITE (with auth digest etc) > Kamailio > INVITE > SIP PROVIDER<br>
<br>
<br>
An attempted solution was made by having Kamailio authenticate using the uac module. However, ideally Kamailio should be mostly transparent and Asterisk should be handling and responding to the 407 Proxy Authentication.<br>
<br>
If there is someone in the Kamailio community that has addressed this situation before, guidance would be much appreciated.<br>
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do you have a failure_route block in kamailio.cfg? Be sure that if 401/407 is received, you just exit the routing block:<br>
<br>
failure_route[abc] {<br>
...<br>
if(t_check_status("401|407")) exit;<br>
...<br>
}<br>
<br>
Then the 401/407 replies will be sent upstream to asterisk.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - <a class="moz-txt-link-freetext" href="http://www.kamailioworld.com" target="_blank">http://www.kamailioworld.com</a></pre>
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