<p dir="ltr">Just be aware, with this any options message from anybody will be answered with ok. </p>
<p dir="ltr">I would prefer to make it specific, at least to be sure that only happens when the source is the asterisk server, and in the right case, because you could get strange behaviors.</p>
<p dir="ltr">Keep testing, and be sure is the right behavior that you need.<br></p>
<div class="gmail_quote">El mar 30, 2014 10:04 PM, "Slava Bendersky" <<a href="mailto:volga629@networklab.ca">volga629@networklab.ca</a>> escribió:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div style="font-size:12pt;font-family:lucida console,sans-serif"><div>Hello Pedro,<br></div><div>I set in main routing section and it I see 200 OK right now. Than you for help.</div><div><br></div><div><br></div><div>
<br></div><div> # account only INVITEs<br> if (is_method("INVITE"))<br> {<br> setflag(FLT_ACC); # do accounting<br> }<br><br> if (is_method("OPTIONS")) {<br>
sl_send_reply("200", "OK");<br> exit;<br> }<br><br></div><div><br></div><div>U 2014/03/30 22:32:47.139646 <a href="http://192.168.10.120:5062" target="_blank">192.168.10.120:5062</a> -> <a href="http://192.168.10.120:5060" target="_blank">192.168.10.120:5060</a><br>
OPTIONS sip:192.168.10.120 SIP/2.0.<br>Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0.<br>Max-Forwards: 70.<br>From: "asterisk" <sip:1300@networklab.loc>;tag=as2cbae229.<br>To: <sip:192.168.10.120>.<br>
Contact: <<a href="http://sip:1300@192.168.10.120:5062" target="_blank">sip:1300@192.168.10.120:5062</a>>.<br>Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc.<br>CSeq: 102 OPTIONS.<br>User-Agent: Asterisk PBX 12.0.0.<br>
Date: Mon, 31 Mar 2014 02:32:47 GMT.<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.<br>Supported: replaces, timer.<br>Content-Length: 0.<br>.<br><br><br>U 2014/03/30 22:32:47.140301 <a href="http://192.168.10.120:5060" target="_blank">192.168.10.120:5060</a> -> <a href="http://192.168.10.120:5062" target="_blank">192.168.10.120:5062</a><br>
SIP/2.0 200 OK.<br>Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK6f8354a0.<br>From: "asterisk" <sip:1300@networklab.loc>;tag=as2cbae229.<br>To: <sip:192.168.10.120>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93f4.<br>
Call-ID: 456f80c06d85d9b34027ccc533855f72@networklab.loc.<br>CSeq: 102 OPTIONS.<br>Server: kamailio (4.1.2 (x86_64/linux)).<br>Content-Length: 0.<br><br></div><div>Slava.<br></div><div><br></div><hr><div style="font-size:12pt;font-style:normal;font-family:Helvetica,Arial,sans-serif;text-decoration:none;font-weight:normal">
<b>From: </b>"Pedro Niño" <<a href="mailto:nino.pedro@gmail.com" target="_blank">nino.pedro@gmail.com</a>><br><b>To: </b>"Kamailio (SER) - Users Mailing List" <<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>><br>
<b>Sent: </b>Sunday, March 30, 2014 10:04:30 PM<br><b>Subject: </b>Re: [SR-Users] message 484<br><div><br></div><p dir="ltr">Ok, I wonder....</p><p dir="ltr">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.</p>
<p dir="ltr">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.</p>
<p dir="ltr">If not, can you explain when are you seeing such behavior? And can run a 'sngrep host <asterisk_IP>' at the Kamailio server?</p><p dir="ltr">Keep me posted.</p><div class="gmail_quote">El mar 30, 2014 8:26 PM, "Slava Bendersky" <<a href="mailto:volga629@networklab.ca" target="_blank">volga629@networklab.ca</a>> escribió:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12pt;font-family:lucida console,sans-serif"><div>Hello Pedro,<br></div><div>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 <br>
</div><div><br></div><div>When Asterisk send reply back this message show up because is no $rU in header line.<br></div><div><br></div><div>I don't know how to correct it that it will full uri format line.<br></div><div>
<br></div><div>voice 192.168.10.120 Auto (No) No A 5060 OK (2 ms) <br>3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline]<br>
<br></div><div><br></div><div><--- SIP read from UDP:<a href="http://192.168.10.120:5060" target="_blank">192.168.10.120:5060</a> ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422<br>
From: "asterisk" <sip:1300@networklab.loc>;tag=as5c659db3<br>To: <sip:192.168.10.120>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2<br>Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc<br>CSeq: 102 OPTIONS<br>
Contact: <<a href="http://sip:1300@192.168.10.120:5062" target="_blank">sip:1300@192.168.10.120:5062</a>>;expires=3600<br>Server: kamailio (4.1.2 (x86_64/linux))<br>Content-Length: 0<br><div><br></div><-------------><br>
--- (9 headers 0 lines) ---<br>Really destroying SIP dialog '4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc' Method: OPTIONS<br><div><br></div><--- SIP read from UDP:<a href="http://192.168.10.120:5060" target="_blank">192.168.10.120:5060</a> ---><br>
SIP/2.0 484 Address Incomplete<br>Via: SIP/2.0/UDP 192.168.10.120:5062;branch=z9hG4bK4c787422<br>From: "asterisk" <sip:1300@networklab.loc>;tag=as5c659db3<br>To: <sip:192.168.10.120>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2<br>
Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee@networklab.loc<br>CSeq: 102 OPTIONS<br>Contact: <<a href="http://sip:1300@192.168.10.120:5062" target="_blank">sip:1300@192.168.10.120:5062</a>>;expires=3600<br>Server: kamailio (4.1.2 (x86_64/linux))<br>
Content-Length: 0<br><div><br></div></div><div><br></div><div> if ($rU==$null)<br> {<br> # request with no Username in RURI<br> sl_send_reply("484","Address Incomplete");<br>
exit;<br> }<br><div><br></div></div><div><br></div><div>Slava.<br></div><div><br></div><hr><div style="font-size:12pt;font-style:normal;font-family:Helvetica,Arial,sans-serif;text-decoration:none;font-weight:normal">
<b>From: </b>"Pedro Niño" <<a href="mailto:nino.pedro@gmail.com" target="_blank">nino.pedro@gmail.com</a>><br><b>To: </b>"Kamailio (SER) - Users Mailing List" <<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>><br>
<b>Sent: </b>Sunday, March 30, 2014 8:30:56 PM<br><b>Subject: </b>Re: [SR-Users] message 484<br><div><br></div><p dir="ltr">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</p>
<p dir="ltr">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.</p><p dir="ltr">Maybe that way we can help.</p><div class="gmail_quote">
El mar 29, 2014 1:59 AM, "Slava Bendersky" <<a href="mailto:volga629@networklab.ca" target="_blank">volga629@networklab.ca</a>> escribió:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div style="font-size:12pt;font-family:lucida console,sans-serif"><div>Hello Everyone,<br></div><div>How to correct message 484 <br></div><div>Is need use txt module to fill string with correct information ?<br></div>
<div><br></div><div><--- SIP read from UDP:<a href="http://192.168.100.145:5060" target="_blank">192.168.100.145:5060</a> ---><br>SIP/2.0 484 Address Incomplete<br>Via: SIP/2.0/UDP 192.168.100.145:5062;branch=z9hG4bK5ec564e6<br>
From: "asterisk" <sip:1300@networklab.loc>;tag=as0a530a8d<br>To: <sip:192.168.100.145>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question.<br>Call-ID: 631e893f75da720865e8468132884367@networklab.loc<br>
CSeq: 102 OPTIONS<br>Contact: <<a href="http://sip:1300@192.168.100.145:5062" target="_blank">sip:1300@192.168.100.145:5062</a>>;expires=3600<br>Server: kamailio (4.1.2 (x86_64/linux))<br>Content-Length: 0<br><div>
<br></div></div><div><br></div><div>Slava.<br></div></div></div><br>_______________________________________________<br> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br> <a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
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