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

[Stefano Sasso]

2010/10/6 Vallimamod ABDULLAH <vallimamod.abdullah at imtelecom.fr>:
> You are right: you should not mix record_route_preset() and record_route().

ok

> Try to replace record_route with record_route preset. And if it does not work, make a ngrep capture on your opensips server to see sip dialog between opensips and asterisk (command line: ngrep -qt -d ethX -W byline port 5060.)

it doesn't work :(
the ACK loops in the opensips, never reaching asterisk.

here a part of ngrep output

U 2010/10/06 14:43:42.733498 94.33.32.xx:55484 -> 192.168.6.130:5060
ACK sip:101 at 77.238.yy.zz:5060 SIP/2.0.
Via: SIP/2.0/UDP
94.33.32.xx:55484;branch=z9hG4bK-d8754z-605d4c7aa5492b3c-1---d8754z-;rport.
Max-Forwards: 70.
Route: <sip:77.238.yy.zz:5060;lr;ftag=931ba062;did=12c.0478d917>.
Contact: <sip:4002 at 94.33.32.xx:55484>.
To: "101"<sip:101 at voip.mydomain.it>;tag=as06487c1d.
From: "4002"<sip:4002 at voip.mydomain.it>;tag=931ba062.
Call-ID: YjZhZTkxYjhiOTA2NjE4NTMzZDk4ZWI1YWM5NmI0M2Y..
CSeq: 2 ACK.

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.
Via: SIP/2.0/UDP 192.168.6.130;branch=z9hG4bK55cc.c8f0d087.2.
Via: SIP/2.0/UDP
94.33.32.xx:55484;received=94.33.32.xx;branch=z9hG4bK-d8754z-605d4c7aa5492b3c-1---d8754z-;rport=55484.
Max-Forwards: 69.
Contact: <sip:4002 at 94.33.32.xx:55484>.
To: "101"<sip:101 at voip.mydomain.it>;tag=as06487c1d.
From: "4002"<sip:4002 at voip.mydomain.it>;tag=931ba062.
Call-ID: YjZhZTkxYjhiOTA2NjE4NTMzZDk4ZWI1YWM5NmI0M2Y..
CSeq: 2 ACK.

U 2010/10/06 14:43:42.737441 192.168.6.130:5060 -> 192.168.6.130:5060
ACK sip:77.238.yy.zz:5060;lr;ftag=931ba062;did=12c.0478d917 SIP/2.0.
Via: SIP/2.0/UDP 192.168.6.130;branch=z9hG4bK55cc.c8f0d087.2.
Via: SIP/2.0/UDP
94.33.32.xx:55484;received=94.33.32.xx;branch=z9hG4bK-d8754z-605d4c7aa5492b3c-1---d8754z-;rport=55484.
Max-Forwards: 69.
Contact: <sip:4002 at 94.33.32.xx:55484>.
To: "101"<sip:101 at voip.mydomain.it>;tag=as06487c1d.
From: "4002"<sip:4002 at voip.mydomain.it>;tag=931ba062.
Call-ID: YjZhZTkxYjhiOTA2NjE4NTMzZDk4ZWI1YWM5NmI0M2Y..
CSeq: 2 ACK.


> Btw, I encourage you to use a public ip on your server if you have the possibility: putting opensips behind nat is *bad* as everybody will tell you ;-)

unfortunately I can't.
the customer would not use more than one ip address, and the servers
are behind a firewall that do DNAT. (all public ip are on external if)

thanks

-- 
Stefano Sasso
http://stefano.dscnet.org/