[OpenSIPS-Users] opensips+asterisk: signalling not working?

[Andrew Pogrebennyk]

On 06.10.2010 16:36, Stefano Sasso wrote:
> nothing happened.
> It still loops (ACKs and BYEs)

Hm, I will have to check in detail what you wrote here.

This ACK should reach the asterisk:
U 2010/10/06 14:43:42.736777 192.168.6.130:5060 -> 77.238.yy.zz:5060
ACK sip:77.238.yy.zz:5060;lr;ftag=931ba062;did=12c.0478d917 SIP/2.0.
...
but then there is another ACK to itself.

Are you doing NAT 77.238.yy.zz to 192.168.6.130 (opensips itself)?
How do you reach the asterisk? I think it should have a mapped routable 
IP address to.


About the correctness of your config, you may remove the record_route() 
from loose_route block which is marked with "even if in most of the 
cases is useless.." comment. You only need this:

	# record routing
	if (!is_method("REGISTER|MESSAGE"))
		record_route_preset("77.238.xx.yy:5060");

IP should be the same as in advertised_address setting. Also add 
force_rport() at the very top of the main route.

Note 1: you do need the advertised_address setting.

Note 2: after removing IPs from domain table you may need to replace
if (!is_uri_host_local())
..
with equivalent check:
if(!uri==myself)

for outbound routing. At least it worked for me.

Anyway the main question is how do you reach the asterisk.

-- 
Sincerely,
Andrew Pogrebennyk