Actually, After reading back the logs: > [Oct 6 10:29:54] WARNING[25602]: chan_sip.c:3805 retrans_pkt: Hanging up > call NjZjMmI2MWRlYmY0YWYwMGVhYTAyNmE0NzU4OWU5YTk. - no reply to our > critical packet (see doc/sip-retransmit.txt).
It is asterisk that is not receiving the ACK so the issue is on your opensips config. Can you make a ngrep trace of an invite to see where is sent the final ACK from opensips ? More precisely, check if the UAC sends the ACK to Opensips' public IP and not the private one. Regards, - vma . On Oct 6, 2010, at 12:22 PM, Stefano Sasso wrote: > 2010/10/6 Vallimamod ABDULLAH <[email protected]>: >> Hi Stefano, > > Hi, > >> Make a sip trace on your asterisk box to see where the ACK is sent. Maybe >> you need to enable nat on asterisk to force it to send the ACK to the >> originating IP and not the IP of the contact field. Have a look at >> http://www.voip-info.org/wiki/view/Asterisk+sip+nat > > now I have nat=yes ; > in the asterisk documentation I read that with nat=yes asterisk > replies directly to the source IP address, ignoring SIP headers. > So, now I assume this is wrong, because the source ip is opensips. > But I can't understand if I must use no, never or route. > > thanks so much, > > -- > Stefano Sasso > http://stefano.dscnet.org/ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
