Thanks for this, I suppose this means the initial INVITE from Asterisk should be adjusted.
Trying a custom trunk so the INVITE r-uri from Asterisk has the provider username but the To header has the number to dial.
SIP/PROVIDERPEER/providerusername!$OUTNUM$
Until today I didn't understand how Asterisk was able to manage r-uri and To header separately.
But I stumbled upon zmonkey whilst searching for "asterisk r-uri".
https://www.zmonkey.org/blog/content/asterisk-sip-request-uri-vs-header
Regards,
Darren
From: Daniel-Constantin Mierla [miconda@gmail.com]
Sent: Monday, 11 May 2015 8:47 PM
To: Darren Campbell (Primar); Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Handling 407 Proxy Authentication, Elastix MT
What is happening then, is the provider sending back another 407?
Normally the Proxy-Authorization header should stay unchanged, but if you change the request uri, it may result in mismatch.
Cheers,
Daniel
On 11/05/15 10:26, Darren Campbell (Primar) wrote:
Thanks, much appreciated.
I'm seeing the Proxy-Authorization from Asterisk in tcpdump. It seems like I've been working against what's already built into Kamailio.
Probably need to tweak some uri's though.
When dialing out, the r-uri is:
sip:mobilenumberhere@exampleip
But the uri part of the Proxy-Authorization in the new INVITE ends up with uri="sip:mobilenumberhere@exampleip"
However, I think it should be showing uri="sip:providerusernamehere@exampleip"
Regards,
Darren
From: sr-users [sr-users-bounces@lists.sip-router.org] on behalf of Daniel-Constantin Mierla [miconda@gmail.com]
Sent: Monday, 11 May 2015 6:07 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Handling 407 Proxy Authentication, Elastix MT
Hello,
On 11/05/15 08:41, Darren Campbell (Primar) wrote:
do you have a failure_route block in kamailio.cfg? Be sure that if 401/407 is received, you just exit the routing block:Hi all
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.
Inbound calls from the SIP PROVIDER work just fine. Have NAT, rtpproxy configured for successful registration and subsequent INVITEs etc.
Experiencing some challenges with the outgoing INVITES, primarily authenticating the outbound INVITEs.
The current situation is this:
Asterisk > INVITE > Kamailio > INVITE > SIP PROVIDER
SIP PROVIDER > 407 Proxy Authenticate > Kamailio > Transaction Cancelled.
Asterisk then plays number unavailable message.
The desired situation is more like this:
Asterisk > INVITE > Kamailio > INVITE > SIP PROVIDER
SIP PROVIDER > 407 Proxy Authenticate > Kamailio > Asterisk
Asterisk > INVITE (with auth digest etc) > Kamailio > INVITE > SIP PROVIDER
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.
If there is someone in the Kamailio community that has addressed this situation before, guidance would be much appreciated.
failure_route[abc] {
...
if(t_check_status("401|407")) exit;
...
}
Then the 401/407 replies will be sent upstream to asterisk.
Cheers,
Daniel
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com