For sake of clarification, the r-uri is not related to the username for
authentication nor to To header uri. They can be completely different.
However, trunk provider might have various requirements in formatting
the r-uri and To uri, they usually say that in the interconnect docs.
Cheers,
Daniel
On 12/05/15 03:11, Darren Campbell (Primar) wrote:
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(a)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(a)lists.sip-router.org] on behalf of
Daniel-Constantin Mierla [miconda(a)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:
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.
do you have a
failure_route block in kamailio.cfg? Be sure that if
401/407 is received, you just exit the routing block:
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