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