[OpenSIPS-Users] openSips - Asterisk and Session Timers: ACK is sent to 192.168.1.10

Bogdan-Andrei Iancu bogdan at voice-system.ro
Mon Oct 25 12:20:47 CEST 2010


Hi,

haven't check your trace, but a fast guess is that you do not fix the 
contact of the 200 OK re-INVITE is no fixed and carries back to asterisk 
a private contact.

So, do fix_nated_contact() for the replies coming from behind a NAT too..

Regards,
Bogdan

    wrote:
> Hi,
>
> My setup:
> - 11.22.33.44 : openSIPS 1.6.3
> - 11.22.33.45 : one of the Asterisk 1.6.2.13 servers
> - 88.77.66.55 : my public ip-address
> - 192.168.1.10 : my local ip-address (NAT)
>
> All is working well except Session Timers where the Re-Invite 
> originates from Asterisk.
>
> I have a SIP trace ( http://pastebin.com/raw.php?i=NRDdaktn ) of a 
> call initiated by a softphone on my pc (192.168.1.10).
> When Asterisk sends the Re-Invite (line 290) my softphone receives 
> this Re-Invite correctly.
> The 100 Trying and 200 OK are also handled as it should.
> But on line 455 you see openSIPS forwarding the ACK to 192.168.1.10 
> instead of 88.77.66.55.
>
> Does anyone know why this isn't working?
> Thanks in advance!
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro




More information about the Users mailing list