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

[Vallimamod ABDULLAH]

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 <vallimamod.abdullah at imtelecom.fr>:
>> 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/
> 
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