27 Dec
2011
27 Dec
'11
2:53 p.m.
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@172.25.249.14
<mailto:sip%3Aasterisk@172.25.249.14>>;tag=as6411602a.
To: <sip:775008@172.25.249.15:5060
<http://sip:775008@172.25.249.15:5060>>;tag=as55ab3180.
Call-ID: 547225391b7828402ecaa03e1dab5a86(a)172.25.249.14
<mailto:547225391b7828402ecaa03e1dab5a86@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@172.25.249.15:5080
<http://sip:775008@172.25.249.15:5080>>.
Remote-Party-ID: "Eus Test" <sip:3999@172.25.249.14
<mailto:sip%3A3999@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@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@172.25.249.14
<mailto:sip%3Aasterisk@172.25.249.14>>;tag=as6411602a.
To: <sip:775008@172.25.249.15:5060
<http://sip:775008@172.25.249.15:5060>>;tag=as55ab3180.
Contact: <sip:asterisk@172.25.249.14
<mailto:sip%3Aasterisk@172.25.249.14>>.
Call-ID: 547225391b7828402ecaa03e1dab5a86(a)172.25.249.14
<mailto:547225391b7828402ecaa03e1dab5a86@172.25.249.14>.
CSeq: 102 ACK.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "asterisk" <sip:asterisk@172.25.249.14
<mailto:sip%3Aasterisk@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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