[SR-Users] ACK From an OK it's being discarded

[Daniel-Constantin Mierla]

Hello,

if the ACK goes through the proxy, then means record routing is used, 
but I see no Record-Route in 200 reply and no Route in ACK. Since there 
is no Record-Route in 200 ok, the ACK has to be sent to the contact 
address from the 200 ok.

Your config snippet from kamailio shows the part of default config where 
record routing is handling -- based on the comments -- since it no 
Route, it is dropped.

Cheers,
Daniel

On 12/26/11 11:03 PM, Lucas Alvarez wrote:
> I have Kamailio 3.2.0 between two asterisk servers, after the call 
> set, one of the kamailio send the OK from the INVITE and the return 
> ACK of that message was discarded. This makes asterisk hangup the call 
> after 5 secs. It's that right?
>
> OK message:
>
> U 172.25.249.15:5060 <http://172.25.249.15:5060> -> 172.25.249.14:5060 
> <http://172.25.249.14:5060>
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 172.25.249.14:5060;branch=z9hG4bK09fc3de6;rport=5060.
> From: "asterisk" <sip:asterisk at 172.25.249.14 
> <mailto:sip%3Aasterisk at 172.25.249.14>>;tag=as6411602a.
> To: <sip:775008 at 172.25.249.15:5060 
> <http://sip:775008@172.25.249.15:5060>>;tag=as55ab3180.
> Call-ID: 547225391b7828402ecaa03e1dab5a86 at 172.25.249.14 
> <mailto:547225391b7828402ecaa03e1dab5a86 at 172.25.249.14>.
> CSeq: 102 INVITE.
> Server: Asterisk PBX 1.8.7.1.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
> INFO, PUBLISH.
> Supported: replaces, timer.
> Contact: <sip:775008 at 172.25.249.15:5080 
> <http://sip:775008@172.25.249.15:5080>>.
> Remote-Party-ID: "Eus Test" <sip:3999 at 172.25.249.14 
> <mailto:sip%3A3999 at 172.25.249.14>>;party=called;privacy=off;screen=no.
> Content-Type: application/sdp.
> Content-Length: 285.
> .
> v=0.
> o=root 2045590031 2045590031 IN IP4 172.25.249.15.
> s=Asterisk PBX 1.8.7.1.
> c=IN IP4 172.25.249.15.
> t=0 0.
> m=audio 11922 RTP/AVP 0 3 8 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
>
> Discarded ACK:
>
> U 172.25.249.14:5060 <http://172.25.249.14:5060> -> 172.25.249.15:5060 
> <http://172.25.249.15:5060>
> ACK sip:775008 at 172.25.249.15:5080 
> <http://sip:775008@172.25.249.15:5080> SIP/2.0.
> Via: SIP/2.0/UDP 172.25.249.14:5060;branch=z9hG4bK6ea5aff6;rport.
> From: "asterisk" <sip:asterisk at 172.25.249.14 
> <mailto:sip%3Aasterisk at 172.25.249.14>>;tag=as6411602a.
> To: <sip:775008 at 172.25.249.15:5060 
> <http://sip:775008@172.25.249.15:5060>>;tag=as55ab3180.
> Contact: <sip:asterisk at 172.25.249.14 
> <mailto:sip%3Aasterisk at 172.25.249.14>>.
> Call-ID: 547225391b7828402ecaa03e1dab5a86 at 172.25.249.14 
> <mailto:547225391b7828402ecaa03e1dab5a86 at 172.25.249.14>.
> CSeq: 102 ACK.
> User-Agent: Asterisk PBX.
> Max-Forwards: 70.
> Remote-Party-ID: "asterisk" <sip:asterisk at 172.25.249.14 
> <mailto:sip%3Aasterisk at 172.25.249.14>>.
> Content-Length: 0.
> .
>
> Kamailio's configuration where the ACK message it's being discarded:
>
>
>   if ( is_method("ACK") ) {
>                                 if ( t_check_trans() ) {
>                                         # no loose-route, but stateful 
> ACK;
>                                         # must be an ACK after a 487
>                                         # or e.g. 404 from upstream server
>                                         t_relay();
>                                         exit;
>                                 } else {
>                                         # ACK without matching 
> transaction ... ignore and discard
>                                         exit;
>                                 }
>                         }
>
>
> It would be ok if I relay the ack even if it didn't match any 
> transaction??
> Any help would be appreciated.
> Regards,
>
> Lucas
>
>
>
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-- 
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda

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