[SR-Users] message 484

Slava Bendersky volga629 at networklab.ca
Mon Mar 31 05:38:58 CEST 2014


Hello Pedro, 
I modified this section to 

if (is_method("OPTIONS") || allow_trusted("$si","$(proto)")) { 
sl_send_reply("200", "OK"); 
exit; 
} 

I am maintaining ACL so should be more tight. 

Slava. 

----- Original Message -----

From: "Pedro Niño" <nino.pedro at gmail.com> 
To: "Kamailio (SER) - Users Mailing List" <sr-users at lists.sip-router.org> 
Sent: Sunday, March 30, 2014 11:20:50 PM 
Subject: Re: [SR-Users] message 484 



Just be aware, with this any options message from anybody will be answered with ok. 

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. 

Keep testing, and be sure is the right behavior that you need. 
El mar 30, 2014 10:04 PM, "Slava Bendersky" < volga629 at networklab.ca > escribió: 



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 at networklab.loc>;tag=as2cbae229. 
To: <sip:192.168.10.120>. 
Contact: < sip:1300 at 192.168.10.120:5062 >. 
Call-ID: 456f80c06d85d9b34027ccc533855f72 at 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 at networklab.loc>;tag=as2cbae229. 
To: <sip:192.168.10.120>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93f4. 
Call-ID: 456f80c06d85d9b34027ccc533855f72 at networklab.loc. 
CSeq: 102 OPTIONS. 
Server: kamailio (4.1.2 (x86_64/linux)). 
Content-Length: 0. 

Slava. 


From: "Pedro Niño" < nino.pedro at gmail.com > 
To: "Kamailio (SER) - Users Mailing List" < sr-users at 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 at networklab.ca > escribió: 

<blockquote>

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 at networklab.loc>;tag=as5c659db3 
To: <sip:192.168.10.120>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 
Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee at networklab.loc 
CSeq: 102 OPTIONS 
Contact: < sip:1300 at 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 at 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 at networklab.loc>;tag=as5c659db3 
To: <sip:192.168.10.120>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b6d2 
Call-ID: 4dc3968e64c3c16c4ad4f2407f2af4ee at networklab.loc 
CSeq: 102 OPTIONS 
Contact: < sip:1300 at 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 at gmail.com > 
To: "Kamailio (SER) - Users Mailing List" < sr-users at 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 at 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 at networklab.loc>;tag=as0a530a8d 
To: <sip:192.168.100.145>;tag=b27e1a1d33761e85846fc98f5f3a7e58.93df ---> This line ins question. 
Call-ID: 631e893f75da720865e8468132884367 at networklab.loc 
CSeq: 102 OPTIONS 
Contact: < sip:1300 at 192.168.100.145:5062 >;expires=3600 
Server: kamailio (4.1.2 (x86_64/linux)) 
Content-Length: 0 


Slava. 

_______________________________________________ 
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sr-users at lists.sip-router.org 
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_______________________________________________ 
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sr-users at 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 at 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 at 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 at lists.sip-router.org 
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 


</blockquote>


_______________________________________________ 
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sr-users at lists.sip-router.org 
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 

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