[SR-Users] message 484

Mon Mar 31 04:34:15 CEST 2014

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. 

----- 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 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: 

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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_______________________________________________ 
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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>

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

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