OpenSER SIP Proxy Users List

Dave Paxton | 25 May 2013 03:18

[SR-Users] question

I have a login with iptel. I am trying to set up jitsi or linphone with someone in argentina. The server sees us both on but we cannot get the thing to work. Each program shows we are online but there is no way to connect. I must be doing something stupid.

Thanks, Dave Paxton

6408 Colonial Drive

Boise, ID 83709

208-570-9755

dpaxton-BUHhN+a2lJ4@public.gmane.org

Skype: dpaxton

<div><div class="WordSection1">
<p class="MsoNormal">I have a login with iptel.&nbsp; I am trying to set up jitsi or linphone with someone in argentina.&nbsp; The server sees us both on but we cannot get the thing to work.&nbsp; Each program shows we are online but there is no way to connect.&nbsp; I must be doing something stupid.<p></p></p>
<p class="MsoNormal"><p>&nbsp;</p></p>
<p class="MsoNormal">Thanks, Dave Paxton<p></p></p>
<p class="MsoNormal">6408 Colonial Drive<p></p></p>
<p class="MsoNormal">Boise, ID 83709<p></p></p>
<p class="MsoNormal">208-570-9755<p></p></p>
<p class="MsoNormal"><a href="mailto:dpaxton@...">dpaxton@...</a><p></p></p>
<p class="MsoNormal">Skype: dpaxton<p></p></p>
<p class="MsoNormal"><p>&nbsp;</p></p>
</div></div>

headers

Mino Haluz | 24 May 2013 17:58

[SR-Users] How does save(location) choose the subscriber

Hi,

if I do save(location) when receiving REGISTER, what is the header which indicates the subscriber for which it will be registered ?

$fu?$au?$tu?

Thanks,

Mino

<div><div dir="ltr">Hi,<div><br></div>
<div>if I do save(location) when receiving REGISTER, what is the header which indicates the subscriber for which it will be registered ?&nbsp;</div>
<div><br></div>
<div>$fu?$au?$tu?</div>
<div><br></div>
<div>Thanks,</div>
<div>Mino</div>
</div></div>

headers

Henning Westerholt | 24 May 2013 11:08

Re: [SR-Users] [sr-dev] "SER Getting Started" - what's the copyright and license?

Am Donnerstag, 23. Mai 2013, 09:56:46 schrieb Olle E. Johansson:
> http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf
> 
> I'm looking for the authors of this document to find out what license it is
> published under. It would be nice to be able to use it and update it
> instead of starting a new "getting started" from scratch.
> 
> Please contact me if you get this message!

Hello Olle,

you've probably contacted them already, for reference this were the authors of 
this document in the past:

Paul Hazlett, phazlett at gmail dot com
Simon Miles, simon at SystemsRM dot co dot uk>
Greger V. Teigre, greger at teigre dot com

Best regards,

Henning Westerholt

headers

Giany | 24 May 2013 11:05

[SR-Users] Calls remain stuck in state 1

Hello,

We are using dipatcher to limit the concurrent number of calls, problem is that from time to time

calls remain stuck in state 1 and it breaks our concurrent limits..I was not able to make a kamailio

log with high debug as it happens randomly. Attached is a tcpdump flow:

Conv.| Time | serverA | Provider |

| | | RemoteEnd |

112 |938.355 | INVITE SDP (g729 g711U GSM X-NSERTPType-100 te...hone-eventRT)

| |(5060) ------------------> (5050) | |

112 |938.357 | 100 Trying| | |

| |(5050) ------------------> (5060) | |

113 |938.420 | INVITE SDP (g711U g729 telephone-eventRTPType-...)|

| |(5050) ------------------> (5060) | |

113 |938.422 | 100 trying -- your call is important to us |

| |(5060) ------------------> (5050) | |

113 |938.422 | INVITE SDP (g711U g729 telephone-eventRTPType-...)|

| |(5060) ------------------> (1416) | |

113 |938.908 | INVITE SDP (g711U g729 telephone-eventRTPType-...)|

| |(5060) ------------------> (1416) | |

113 |939.908 | INVITE SDP (g711U g729 telephone-eventRTPType-...)|

| |(5060) ------------------> (1416) | |

113 |941.906 | INVITE SDP (g711U g729 telephone-eventRTPType-...)|

| |(5060) ------------------> (1416) | |

113 |945.907 | INVITE SDP (g711U g729 telephone-eventRTPType-...)|

| |(5060) ------------------> (1416) | |

112 |948.358 | CANCEL | | |SIP Request

| |(5060) <-------------------------------------- (61016) |

112 |948.359 | CANCEL | | |SIP Request

| |(5060) ------------------> (5050) | |

112 |948.359 | 200 canceling | |SIP Status

| |(5060) --------------------------------------> (61016) |

112 |948.359 | 487 Request Terminated | |SIP Status

| |(5050) ------------------> (5060) | |

112 |948.359 | 200 OK | | |SIP Status

| |(5050) ------------------> (5060) | |

112 |948.359 | ACK | | |SIP Request

| |(5060) ------------------> (5050) | |

112 |948.360 | 487 Request Terminated | |SIP Status

| |(5060) --------------------------------------> (61016) |

113 |948.360 | CANCEL | | |SIP Request

| |(5050) ------------------> (5060) | |

113 |948.360 | 200 canceling | |SIP Status

| |(5060) ------------------> (5050) | |

112 |948.365 | ACK | | |SIP Request

| |(5060) <-------------------------------------- (61016) |

dialog:: hash=136:689416016 state:: 1 ref_count:: 3 timestart:: 0 timeout:: 0 callid:: 745eed805cad6b9a3e1727d169cf3461 <at> serverA:5050 from_uri:: sip:fromnumber <at> serverA:5050
from_tag:: as53f6bee4 caller_contact:: sip:fromnumber <at> serverA:5050 caller_cseq:: 102 caller_route_set:: caller_bind_addr:: udp:serverA:5060
callee_bind_addr:: to_uri:: sip:internalnr <at> serverA:5060
to_tag:: callee_contact:: callee_cseq:: callee_route_set::
As you see the remoteEnd does not answer at all to this request (due to network issue most likely)and the provider sends a CANCEL after approx 3 seconds.From what I see the INVITE that is sent from asterisk towards kamailio remains stuck (938.420).We are using Kamailio 3.1.6. Any idea what could be the reason for this?
Thank you.

<div><div>
<div>Hello,</div>
<div><br></div>
<div>We are using dipatcher to limit the concurrent number of calls, problem is that from time to time</div>
<div>&nbsp;calls remain stuck in state 1 and it breaks our concurrent limits..I was not able to make a kamailio</div>
<div>log with high debug as it happens randomly. Attached is a tcpdump flow:</div>
<div><br></div>
<div><div>
<div>Conv.| Time &nbsp; &nbsp;| serverA &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | Provider &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | RemoteEnd &nbsp; &nbsp; | &nbsp; &nbsp;
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;&nbsp;</div>
<div>112 &nbsp;|938.355 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; INVITE SDP (g729 g711U GSM X-NSERTPType-100 te...hone-eventRT) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(5050) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>112 &nbsp;|938.357 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; 100 Trying| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5050) &nbsp; ------------------&gt; &nbsp;(5060) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>113 &nbsp;|938.420 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; INVITE SDP (g711U g729 telephone-eventRTPType-...)|</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5050) &nbsp; ------------------&gt; &nbsp;(5060) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>113 &nbsp;|938.422 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; 100 trying -- your call is important to us &nbsp; &nbsp;
 &nbsp; &nbsp;|</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(5050) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>113 &nbsp;|938.422 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; INVITE SDP (g711U g729 telephone-eventRTPType-...)|</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(1416) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>113 &nbsp;|938.908 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;
 INVITE SDP (g711U g729 telephone-eventRTPType-...)|&nbsp;</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(1416) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>113 &nbsp;|939.908 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; INVITE SDP (g711U g729 telephone-eventRTPType-...)|</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(1416) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>113
 &nbsp;|941.906 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; INVITE SDP (g711U g729 telephone-eventRTPType-...)|</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(1416) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>113 &nbsp;|945.907 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; INVITE SDP (g711U g729 telephone-eventRTPType-...)|</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(1416) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>112 &nbsp;|948.358 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; CANCEL &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP Request</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; &lt;-------------------------------------- &nbsp;(61016) &nbsp;|</div>
<div>112 &nbsp;|948.359 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; CANCEL &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP Request</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp;
 &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(5050) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>112 &nbsp;|948.359 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; 200 canceling &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP Status</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; --------------------------------------&gt; &nbsp;(61016) &nbsp;|</div>
<div>112 &nbsp;|948.359 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; 487 Request Terminated &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP
 Status</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5050) &nbsp; ------------------&gt; &nbsp;(5060) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>112 &nbsp;|948.359 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; 200 OK &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP Status</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5050) &nbsp; ------------------&gt; &nbsp;(5060) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>112 &nbsp;|948.359 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; ACK &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP Request</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(5050) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>112 &nbsp;|948.360 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; 487 Request Terminated &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP Status</div>
<div>&nbsp; &nbsp;
 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; --------------------------------------&gt; &nbsp;(61016) &nbsp;|</div>
<div>113 &nbsp;|948.360 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; CANCEL &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP Request</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5050) &nbsp; ------------------&gt; &nbsp;(5060) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>113 &nbsp;|948.360 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; 200 canceling &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |
 &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP Status</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; ------------------&gt; &nbsp;(5050) &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |</div>
<div>112 &nbsp;|948.365 &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; ACK &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |SIP Request</div>
<div>&nbsp; &nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp; |(5060) &nbsp; &lt;-------------------------------------- &nbsp;(61016) &nbsp;|</div>
<div><br></div>
<div><br></div>
<div>dialog::  hash=136:689416016
        state:: 1
        ref_count:: 3
        timestart:: 0
        timeout:: 0
        callid:: 745eed805cad6b9a3e1727d169cf3461 <at> <span>serverA</span>:5050
        from_uri:: sip:<span>fromnumber</span><span> <at> serverA:5050</span><br>        from_tag:: as53f6bee4
        caller_contact:: sip:fromnumber <at> serverA:5050
        caller_cseq:: 102
        caller_route_set::
        caller_bind_addr:: udp:<span>serverA</span><span>:5060</span><br>        callee_bind_addr::
        to_uri:: sip:internalnr <at> <span>serverA</span><span>:5060</span><br>        to_tag::
        callee_contact::
        callee_cseq::
        callee_route_set::<br>As you see the remoteEnd does not answer at all to this request (due to network issue most likely)and the provider sends a CANCEL after approx 3 seconds.From what I see the INVITE that is sent from asterisk towards kamailio
 remains stuck (<span>938.420).</span>We are using Kamailio 3.1.6. <span>Any idea what could be the reason for this? </span><span><br></span><span>Thank you.</span>
</div>
</div></div>
</div></div>

headers

Rupayan Dutta | 23 May 2013 13:24

[SR-Users] kamailio 4.0.1 websocket support

Dear All,

I have build and installed kamailio 4.0.1 from source(tarball release) in Cent Os 5.8 in i386 machine.I can start kamailio but when i want to connect kamailio using SipMl5 client which uses sip over websocket the client can't connect with the server.I am attaching my generated kamailio.cfg file with this mail.Is there any additional configuration needed for websocket support in kamailio 4.0.1?I have located the example websocket.cfg file in kamailio source but totally confused what modification to be done in the end of the file(e.g Main SIP request routing logic blocks).

Note: I moified modules.lst and include db_mysql auth auth_db sl nathelper tm xhttp tls msrp websocket modules

Attachment (kamailio.cfg): application/octet-stream, 20 KiB

Attachment (websocket.cfg): application/octet-stream, 11 KiB

<div><div dir="ltr">
<div>
<div>Dear All,<br>
</div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I have build and installed kamailio 4.0.1 from source(tarball release) in Cent Os 5.8 in i386 machine.I can start kamailio but when i want to connect kamailio using SipMl5 client which&nbsp; uses sip over websocket the client can't connect with the server.I am attaching my generated kamailio.cfg file with this mail.Is there&nbsp; any additional configuration needed for websocket support in kamailio 4.0.1?I have located the example websocket.cfg file in kamailio source but totally confused what modification to be done in the end of the file(e.g Main SIP request routing logic&nbsp; blocks)&#8203;&#8203;.<br><br>
</div>Note: I moified modules.lst and include db_mysql auth auth_db sl nathelper tm xhttp tls msrp websocket modules<br>
</div></div>

headers

אורן אברהם | 23 May 2013 17:49

[SR-Users] add_path() and save()

Dear list.

when using registrar module with path support does the following code work:

add_path();

save();

or maybe the path is added only for outgouing messages and the save() won't be affected ?

<div><div dir="rtl">
<div dir="ltr">Dear list.</div>
<div dir="ltr"><br></div>
<div dir="ltr">when using registrar module with path support does the following code work:</div>
<div dir="ltr">add_path();</div>
<div dir="ltr">save();</div>
<div dir="ltr"><br></div>
<div dir="ltr">or maybe the path is added only for outgouing messages and the save() won't be affected ?</div>
<div dir="ltr"><br></div>
<div dir="ltr">
<br>
</div>
</div></div>

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Julia | 23 May 2013 12:21

[SR-Users] topoch and xHTTP

Hello,

I am testing the xHTTP module on kamailio-3.3 with topology hiding (topoch) and

I get parsing error for each received HTTP message:

/usr/local/sbin/kamailio[2937]: ERROR: <core> [parser/parse_from.c:60]: ERROR:parse_from_header: bad msg or missing FROM header
/usr/local/sbin/kamailio[2937]: ERROR: topoh [topoh_mod.c:215]: cannot parse FROM header

Is it configuration issue?

Thanks,

Julia.

<div>

<div class="Section1">

<p class="MsoNormal"><span>Hello,<p></p></span></p>

<span>I am testing the xHTTP module on kamailio-3.3 with topology hiding (topoch)</span><span> </span><span>and <p></p></span><span>I get parsing error for each received HTTP message: &nbsp;<p></p></span>

<p class="MsoNormal"><span><p>&nbsp;</p></span></p>

<div>

<p class="MsoNormal"><span>/usr/local/sbin/kamailio[2937]: ERROR: &lt;core&gt;
[parser/parse_from.c:60]: ERROR:parse_from_header: bad msg or missing FROM
header<br>
/usr/local/sbin/kamailio[2937]: ERROR: topoh [topoh_mod.c:215]: cannot parse
FROM header<p></p></span></p>

<p class="MsoNormal"><span><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span>Is it configuration issue?<p></p></span></p>

<p class="MsoNormal"><span><p>&nbsp;</p></span></p>

<p class="MsoNormal"><span>Thanks,<p></p></span></p>

<p class="MsoNormal"><span>Julia.<p></p></span></p>

</div>

</div>

</div>

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Victor V. Kustov | 23 May 2013 12:19

[SR-Users] Radius and FLT_ACCFAILED

Hello!

I do radius accounting. When I imitate call fail, BYE/CANCEL messages
are loose, account stop packets not sending. I see FLT_ACCFAILED flag,
can I set it for acc_radius module?

--
 WBR, Victor
  JID: coyote@...
  JID: coyote@...
  I use FREE operation system: 3.9.2-calculate GNU/Linux

headers

Richard Brady | 23 May 2013 12:12

[SR-Users] Via header, branch parameter and syn_branch

Hi folks

The syn_branch global parameter results in the use of a "synonym"
branch parameter in the Via header for statelessly forwarded requests
as a performance optimisation.

This was originally done by setting branch=0 which, while not strictly
compliant with 3261 (8.1.1.7 and 16.6 item 8), would not cause any
problems with a downstream proxy or UA compliant to either 3261 or
2543 since it does not contain the cookie and is therefore
self-evidently not guaranteed to be unique.

Then a patch was added (commit ebb3b08) to change it from branch=0 to
branch=z9hG4bKcydzigwkX which includes the cookie, thus declaring
itself to be unique when it clearly is not.

This takes the optimisation too far as it deliberately misleads
downstream proxies with regard to the uniqueness of the parameter and
breaks the mechanism in 17.2.3 for identifying unreliable (non-unique)
branch parameters for transaction matching which should then fall back
to header inspection but does not. The result is that unrelated
requests can be identified as duplicates of each other.

Also note 3261 section 16.11 on Stateless Proxy behaviour: "The
requirement for unique branch IDs across space and time applies to
stateless proxies as well."

Has the commit above introduced a bug and should it be reverted?

And should the next major release have a default of syn_branch=0?
Since with syn_branch=1 the branch=0 version has been known to cause
interop issues in the past (see below) and I can confirm the
branch=z9hG4bKcydzigwkX version causes interop issues in the present.

Regards,
Richard

See also these related threads:

[SR-Users] Via header branch parameter in ACK message not unique
http://lists.sip-router.org/pipermail/sr-users/2011-December/071207.html

[SR-Users] "branch" tag in the "Via" header of the ACK message for the re-INVITE
http://lists.kamailio.org/pipermail/sr-users/2012-November/075560.html

[Serusers] branch parameter
http://lists.sip-router.org/pipermail/sr-users/2007-February/060147.html

[SR-Users] about syn_branch
http://lists.sip-router.org/pipermail/sr-users/2011-March/067732.html

[SR-Users] Via branch parameter in end2end ACK
http://lists.sip-router.org/pipermail/sr-users/2011-May/068627.html

[SR-Users] Broken Via/reply-matching for natping OPTION
http://lists.sip-router.org/pipermail/sr-users/2011-April/068340.html

[Serusers] ACK message Via field branch=0 problem
http://lists.sip-router.org/pipermail/sr-users/2004-July/037731.html

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Andrew Pogrebennyk | 23 May 2013 11:36

[SR-Users] auth module Readme on website has missing ToC

Hi,
please check the link
http://kamailio.org/docs/modules/stable/modules/auth.html - the Table of
Contents is missing, there's only examples list.

Andrew

headers

אורן אברהם | 23 May 2013 02:35

[SR-Users] Websockets and outbound config file

Dear list

Does someone has a working config file for a registrar + outbound edge proxy on the same kamailio server instance that works ? (like the /examples/websocket.cfg but updated to use outbound & path)

thanks...

<div><div dir="rtl">
<div dir="ltr">Dear list</div>
<div dir="ltr"><br></div>
<div dir="ltr">Does someone has a working config file for a registrar + outbound edge proxy on the same kamailio server instance &nbsp;that works ? (like the /examples/websocket.cfg but updated to use outbound &amp; path)&nbsp;</div>
<div dir="ltr"><br></div>
<div dir="ltr">thanks...</div>
</div></div>

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