Hello Pedro, I set in main routing section and it I see 200 OK right now. Than you for help.
# account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting }
if (is_method("OPTIONS")) { sl_send_reply("200", "OK"); exit; }
U 2014/03/30 22:32:47.139646 192.168.10.120:5062 -> 192.168.10.120:5060 OPTIONS sip:192.168.10.120 SIP/2.0. Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0. Max-Forwards: 70. From: "asterisk" sip:1300@networklab.loc;tag=as2cbae229. To: sip:192.168.10.120. Contact: sip:1300@192.168.10.120:5062. Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc. CSeq: 102 OPTIONS. User-Agent: Asterisk PBX 12.0.0. Date: Mon, 31 Mar 2014 02:32:47 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. .
U 2014/03/30 22:32:47.140301 192.168.10.120:5060 -> 192.168.10.120:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0. From: "asterisk" sip:1300@networklab.loc;tag=as2cbae229. To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.93f4. Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc. CSeq: 102 OPTIONS. Server: kamailio (4.1.2 (x86_64/linux)). Content-Length: 0.
Slava.
----- Original Message -----
From: "Pedro Niño" nino.pedro@gmail.com To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Sent: Sunday, March 30, 2014 10:04:30 PM Subject: Re: [SR-Users] message 484
Ok, I wonder....
If this is a message you're seeing at the asterisk server, it may be related to the qualify=yes or qualify=Number parameter in the peer, at sip.conf.
If it's right, then you can modify at 2 places: one, by disabling qualify. (qualify=no) at asterisk, or the other by configuring Kamailio to answer a 200 "OK" message when the message comes from the asterisk server.
If not, can you explain when are you seeing such behavior? And can run a 'sngrep host <asterisk_IP>' at the Kamailio server?
Keep me posted. El mar 30, 2014 8:26 PM, "Slava Bendersky" < volga629@networklab.ca > escribió:
Hello Pedro, So test case is simple. asterisk is play voicemail role and kamailio is gateway. Asterisk peer is setup and working. I think message SIP/2.0 484 Address Incomplete is some from kamailio
When Asterisk send reply back this message show up because is no $rU in header line.
I don't know how to correct it that it will full uri format line.
voice 192.168.10.120 Auto (No) No A 5060 OK (2 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]
<--- SIP read from UDP: 192.168.10.120:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.10.120:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc' Method: OPTIONS
<--- SIP read from UDP: 192.168.10.120:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422 From: "asterisk" sip:1300@networklab.loc;tag=as5c659db3 To: sip:192.168.10.120;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.10.120:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }
Slava.
From: "Pedro Niño" < nino.pedro@gmail.com > To: "Kamailio (SER) - Users Mailing List" < sr-users@lists.sip-router.org > Sent: Sunday, March 30, 2014 8:30:56 PM Subject: Re: [SR-Users] message 484
I think this is the correct behavior, as asterisk server is complaining about the address/request not containing all the necesary data to process the message
Can you please elaborate with a bit more of detail? Also can use tools like sngrep, tcpdump (or wireshark) to have a better view of the complete call flow.
Maybe that way we can help. El mar 29, 2014 1:59 AM, "Slava Bendersky" < volga629@networklab.ca > escribió:
<blockquote>
Hello Everyone, How to correct message 484 Is need use txt module to fill string with correct information ?
<--- SIP read from UDP: 192.168.100.145:5060 ---> SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6 From: "asterisk" sip:1300@networklab.loc;tag=as0a530a8d To: sip:192.168.100.145;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. Call-ID: 631e893f75da720865e8468132884367@networklab.loc CSeq: 102 OPTIONS Contact: < sip:1300@192.168.100.145:5062 >;expires=3600 Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0
Slava.
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_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
</blockquote>
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users