Re: [SR-Users] early media: How to change to tag

[Daniel-Constantin Mierla]

Hello,

On 3/29/12 1:14 AM, Andreas Granig wrote:

Hi,

On 03/28/2012 06:37 PM, Iñaki Baz Castillo wrote:

2012/3/28 Min Wang:

In order to properly proxy the msg to GW1, Kamailio seems need to change the
to tag from B to A.

Totally wrong. Multiple (early-)dialogs are 100% valid according to
RFC 3261. If you find some SIP device failing when it receives
multiple 180/183/200 responses with different To-tag, then drop it

I recently learned that for example Siemens switches implement "Request
Disposition: no-fork" defined in http://www.ietf.org/rfc/rfc3841.txt,
and if for some reason you decide to fork nonetheless on your side,
you'd probably want to do something about the different to-tags
(although you're violating against that specific RFC then). No idea how
such a device would react to not getting a to-tag at all by stripping it
out as Klaus suggested in another response, but at least that Siemens
switch doesn't bail out on getting different to-tags in provisional replies.
I fully agree that devices not supporting multiple to-tags in 
provisional replies should be dropped, it will make life easier for 
everyone.


Regarding hacks to make an workaround, in many cases I saw devices 
requiring to-tag in 183. In that case, a solution can be dropping 183 
from reply route and send 180 instead with t_reply(...).


Cheers,
Daniel

--
Daniel-Constantin Mierla
Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany
http://www.asipto.com/index.php/kamailio-advanced-training/


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Re: [SR-Users] Update of wiki write access

[Daniel-Constantin Mierla]

Hello,

anyone has experiences with captcha vs recaptcha ( http://recaptcha.net/).

There is a plugin for dokuwiki 
(http://www.dokuwiki.org/plugin:recaptcha) -- my interest is how strong 
the system is to defend against bot spammers, rather than other good 
faith things (i.e., helping scanning books for free).


Cheers,
Daniel

On 3/26/12 10:24 AM, Daniel-Constantin Mierla wrote:

Hello,

just to let you know that over the weekend the ACL (user 
authentication) for write access to wiki portals of the project has 
been turned on.


Anyone that wants to write to the wiki has to create herself/himself 
an account. The reason for requiring user authentication is weakness 
of the captha system used for anonymous posting -- during the past 
weeks lot of wiki pages were updated by spam bots.


If anyone is aware of a stronger captcha-like system to be used for 
anonymous posting, write back here. It has to work with dokuwiki.


Cheers,
Daniel



--
Daniel-Constantin Mierla
Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany
http://www.asipto.com/index.php/kamailio-advanced-training/

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Re: [SR-Users] AVPs in timer-route

[Daniel-Constantin Mierla]

Hello,

On 3/21/12 6:05 PM, Klaus Darilion wrote:

On 20.03.2012 23:36, Daniel-Constantin Mierla wrote:

Hello,

On 3/20/12 8:13 PM, Klaus Darilion wrote:

Hi!

I wonder what happens to AVPs defined in a route which is executed by
rtimer?

It seems (tests on Kamailio 1.5) that the AVPs, once they are defined,
they life forever. This would be fine for my application.

Are there any issues when using AVPs in timer routes?

there is no issue using avps in rtimer routes. Those routes process a
faked sip message, which does not change, just the internal id is
increased. AVPs live for ever because they are cleaned up after a
received messages is processed, but in this case there is no received
context. It can be added to reset avps in rtimer module.

On the other hand, avps are usually useful in context of a real sip
message. Why you need them in a timer route?


I use rtimer module (backported from 1.5 to a quite extended openser 
1.3) to dump statistics in the timer route. The first time the timer 
route is executed I define some AVP-arrays (e.g. IP addresses of 
servers) over wich I loop in a while loop to dump the respective 
statistics.
Then it is working fine with current implementation - the avps 
associated to the fake message given to timer route are not deleted at 
all, avp_delete() from avpops can be used to do it on purpose if one wants.


Btw, if you load the lost from a db table, if you use sqlops/sql_query() 
and don't free the result, it is available again and again -- it is 
another way to keep some data in private memory and be able to loop over 
it many times.


Cheers,
Daniel

--
Daniel-Constantin Mierla
Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany
http://www.asipto.com/index.php/kamailio-advanced-training/


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[SR-Users] Missing "Route" in ACK or BYE request

[Hervé Cochet]

Hello,

I used kamailio to handle SIP line from a provider using uacreg table.
Everything was working fine till they add a proxy which do not set (or
not propagate) route header in ACK or BYE request.

So my routing logic is broken in the "in-dialog" transaction mode,
since the loose_route return FALSE.

If I try to force a t_relay after the loose_route, my message may not
be routing properly if for example my UAC was not connected with the
standard 5060 port or use a different protocol like TCP (It works if
both are in UDP and use 5060 port).
Is there a way to retrieve the record-route previously sets for this
session in order to route this message properly ?



Another problem is that the dialog module do not match the BYE
transaction, the "did" variable is missing since the route is not there
but it should match the request with SIP matching (dlg_match_mode is
set to 1).

For example my dialog parameters are:

dialog::  hash=1214:1340198900
    state:: 3
    timestart:: 1333006717
    timeout:: 80537132
    callid:: 16260-ci-30dd2867-797784...@my.sip.provider
    from_uri:: sip:0367023...@my.sip.provider;user=phone
    from_tag:: 16260-GI-30dd2868-17bb342c3
    caller_contact:: sip:13.12.14.20:5060
    caller_cseq:: 815264875
    caller_route_set:: 
    caller_bind_addr:: udp:130.120.140.131:5060
    to_uri:: sip:0974711672@13.12.14.17;user=phone
    to_tag:: 1878467993
    callee_contact:: sip:0974711672@96.57.249.78:1024
    callee_cseq:: 815264875
    callee_route_set:: 
    callee_bind_addr:: udp:130.120.140.131:5060

The BYE request:
BYE sip:0974711672@96.57.249.78:1024 SIP/2.0

Call-ID: 16260-ci-30dd2867-797784...@my.sip.provider

CSeq: 815264876 BYE

From: "0033482531303"
;tag=16260-GI-30dd2868-17bb342c3


Max-Forwards: 28

Record-Route: 

To: ;tag=1878467993

Via: SIP/2.0/UDP 13.12.14.20:5060;branch=z9hG4bK-LNVP-1196a924-5f1f495a

Reason: q.850;cause=16

User-Agent: Cirpack/v4.42a (gw_sip)

Content-Length: 0

Why does this BYE request is not matched by the dialog modules using
SIP parameters (Call-ID, uri, tag seems correct) ?


Hervé



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[SR-Users] Reload lua script?

[Javier Gallart]

Hello all

I'm starting to play with lua scripting in Kamailio. It's really cool, some
complex stuff in the script becomes really easy this way. The script is
becoming quite big, therefore I'd like to use lua_run instead of
lua_dofile. My doubt is pretty  basic: if i load a script in memory at
startup via the "load" parameter and the script is modifed afterwards, is
there any way to reload the module so the script is reread?

Thanks in advance

Regards

Javi
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Re: [SR-Users] branches usage

[Carsten Bock]

Hi,

you need the "append_branch" only, if you modify the request URI in
your config (e.g. by doing a lookup()) and you want to have another
branch (e.g. parallel forking to voicebox).

Carsten

2012/3/28 Uri Shacked :
> OK...
> It worked great... and, no append_branch() was needed... only t_on_branch
> with the main branch route.
> When will i need to use the append_branch?
> BR,
> Uri
>
> On Wed, Mar 28, 2012 at 4:00 PM, Uri Shacked  wrote:
>>
>> Hi,
>>
>> Thanks, this was very helpful for understanding.
>>
>> Still let’s see if i got it right:
>>
>> I get the INVITE.
>>
>> Do whatever I do on the main route.
>>
>> Then do:
>>
>> “append_branch();”
>>
>> “changes….changes…..;”
>>
>>  Now before the t_relay do t_on_branch(name of branch).
>>
>> Get the reply (301 in my case)
>>
>> And from here I am back to the original request?
>>
>> maybe a stupid question now but - I must use the append_branch()  right
>> :-)?
>>
>> BR,
>>
>> Uri
>>
>>
>>
>> Hi,
>>
>> On 03/27/2012 04:46 PM, Uri Shacked wrote:
>> > In my case i need to change the header and then send it.
>> >
>> > Here the case works if i make the changes after i sent the invite and
>> > got the reply. As i know the t_on_branch will work after the invite is
>> > sent... am i wrong?
>> >
>> > There are many questions to ask about the branches... when do they work
>> > > in a serial manner, when parallel... where can i find it documented best?
>>
>> Actually it's quite simple. If a request comes in and you modify it in a
>> normal route (e.g. calling rtpproxy_offer(), uac_replace_from(), adding
>> additional headers and stuff like that) and you send it out using e.g.
>> t_relay(), then in a serial scenario when you re-enter via a
>> failure-route, these changes will still be there (which is not always
>> what you want).
>>
>> On the other hand, if you register a branch route via t_on_branch() and
>> do your changes there instead of prior to calling t_relay(), then these
>> changes from branch route will be rolled back, and you can do your
>> changes easily again in the same or another branch route starting with
>> your original message. This is important for example in serial call
>> forward scenarios, where the first destination might require an
>> rtpproxy, but the second doesn't, or when the first destination requires
>> different custom headers compare to the second.
>>
>> Basically the same applies to parallel scenarios. If you modify your
>> message before calling lookup() and you've multiple contacts registered
>> for your destination, then your changes will apply for all branches
>> which are created by that. However if you move your modifications to a
>> branch route, then you can work on your messages individually, depending
>> on the branch.
>>
>> Hope this clarifies it a bit.
>>
>> Andreas
>>
>
>
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>



-- 
Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany

http://www.ng-voice.com
mailto:cars...@ng-voice.com

Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220

Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284

Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/

-- 
Meet ng-voice at LinuxTag 2012 in Berlin - May 23rd - 26th, 2012. Save the 
date!

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Re: [SR-Users] Update of wiki write access

[Olle E. Johansson]

29 mar 2012 kl. 09:45 skrev Daniel-Constantin Mierla:

> Hello,
> 
> anyone has experiences with captcha vs recaptcha ( http://recaptcha.net/).
I'm using recaptcha on a couple of sites. It works fine for me.

On the downside, it dies with IPv6...
Hopefully google is going to fix that, at least to the World IPv6 Launch.
/O
> 
> There is a plugin for dokuwiki (http://www.dokuwiki.org/plugin:recaptcha) -- 
> my interest is how strong the system is to defend against bot spammers, 
> rather than other good faith things (i.e., helping scanning books for free).
> 
> Cheers,
> Daniel
> 
> On 3/26/12 10:24 AM, Daniel-Constantin Mierla wrote:
>> Hello, 
>> 
>> just to let you know that over the weekend the ACL (user authentication) for 
>> write access to wiki portals of the project has been turned on. 
>> 
>> Anyone that wants to write to the wiki has to create herself/himself an 
>> account. The reason for requiring user authentication is weakness of the 
>> captha system used for anonymous posting -- during the past weeks lot of 
>> wiki pages were updated by spam bots. 
>> 
>> If anyone is aware of a stronger captcha-like system to be used for 
>> anonymous posting, write back here. It has to work with dokuwiki. 
>> 
>> Cheers, 
>> Daniel 
>> 
> 
> -- 
> Daniel-Constantin Mierla
> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany
> 
> http://www.asipto.com/index.php/kamailio-advanced-training/
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---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden




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[SR-Users] how to configure Kamailio in ECHO mode

[Fomitchev, AndreiX]

Hi,

For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. 
SIP server
-
Intel Corporation SAS (French simplified joint stock company)
Registered headquarters: "Les Montalets"- 2, rue de Paris, 
92196 Meudon Cedex, France
Registration Number:  302 456 199 R.C.S. NANTERRE
Capital: 4,572,000 Euros

This e-mail and any attachments may contain confidential material for
the sole use of the intended recipient(s). Any review or distribution
by others is strictly prohibited. If you are not the intended
recipient, please contact the sender and delete all copies.
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Re: [SR-Users] how to configure Kamailio in ECHO mode

[Fomitchev, AndreiX]

Oups, sorry, bad manipulation:
I want to configure Kamailio in echo mode, i.e. SIP server repeats the input 
audio into the output, so I can hear myself with a very small amount of delay...

Is it possible? If so, how?

Thank you for your help.

Best regards,

From: Fomitchev, AndreiX
Sent: Thursday, March 29, 2012 1:34 PM
To: 'sr-users@lists.sip-router.org'
Subject: how to configure Kamailio in ECHO mode

Hi,

For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. 
SIP server
-
Intel Corporation SAS (French simplified joint stock company)
Registered headquarters: "Les Montalets"- 2, rue de Paris, 
92196 Meudon Cedex, France
Registration Number:  302 456 199 R.C.S. NANTERRE
Capital: 4,572,000 Euros

This e-mail and any attachments may contain confidential material for
the sole use of the intended recipient(s). Any review or distribution
by others is strictly prohibited. If you are not the intended
recipient, please contact the sender and delete all copies.
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Re: [SR-Users] how to configure Kamailio in ECHO mode

[Alex Balashov]

Can you elaborate on what that means?

If you are talking about acoustic echo, it's not possible. Kamailio is a proxy 
/ SIP server, it does not handle media or answer calls or anything like that.

--
Alex Balashov - Principal 
Evariste Systems LLC 
235 E Ponce de Leon Ave 
Suite 106
Atlanta, GA 30030 
Tel: +1-678-954-0670 
Fax: +1-404-961-1892 
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

"Fomitchev, AndreiX"  wrote:

>Hi,
>
>For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. 
>SIP server
>-
>Intel Corporation SAS (French simplified joint stock company)
>Registered headquarters: "Les Montalets"- 2, rue de Paris, 
>92196 Meudon Cedex, France
>Registration Number:  302 456 199 R.C.S. NANTERRE
>Capital: 4,572,000 Euros
>
>This e-mail and any attachments may contain confidential material for
>the sole use of the intended recipient(s). Any review or distribution
>by others is strictly prohibited. If you are not the intended
>recipient, please contact the sender and delete all copies.
>
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Re: [SR-Users] how to configure Kamailio in ECHO mode

[Alex Balashov]

No, it's not possible.  I think you have the wrong idea about what Kamailio is. 
 Kamailio neither receives nor produces audio at all.

--
Alex Balashov - Principal 
Evariste Systems LLC 
235 E Ponce de Leon Ave 
Suite 106
Atlanta, GA 30030 
Tel: +1-678-954-0670 
Fax: +1-404-961-1892 
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

"Fomitchev, AndreiX"  wrote:

>Oups, sorry, bad manipulation:
>I want to configure Kamailio in echo mode, i.e. SIP server repeats the input 
>audio into the output, so I can hear myself with a very small amount of 
>delay...
>
>Is it possible? If so, how?
>
>Thank you for your help.
>
>Best regards,
>
>From: Fomitchev, AndreiX
>Sent: Thursday, March 29, 2012 1:34 PM
>To: 'sr-users@lists.sip-router.org'
>Subject: how to configure Kamailio in ECHO mode
>
>Hi,
>
>For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. 
>SIP server
>-
>Intel Corporation SAS (French simplified joint stock company)
>Registered headquarters: "Les Montalets"- 2, rue de Paris, 
>92196 Meudon Cedex, France
>Registration Number:  302 456 199 R.C.S. NANTERRE
>Capital: 4,572,000 Euros
>
>This e-mail and any attachments may contain confidential material for
>the sole use of the intended recipient(s). Any review or distribution
>by others is strictly prohibited. If you are not the intended
>recipient, please contact the sender and delete all copies.
>
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Re: [SR-Users] how to configure Kamailio in ECHO mode

[Fomitchev, AndreiX]

Hi Alex,

Thank you for your response. Please, could you tell me what I need to setup a 
SIP “echo service”? I am new in SIP domain :/

Regards,

From: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex Balashov
Sent: Thursday, March 29, 2012 1:38 PM
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users 
Mailing List
Subject: Re: [SR-Users] how to configure Kamailio in ECHO mode

No, it's not possible. I think you have the wrong idea about what Kamailio 
is. Kamailio neither receives nor produces audio at all.

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

"Fomitchev, AndreiX" 
mailto:andreix.fomitc...@intel.com>> wrote:
Oups, sorry, bad manipulation:
I want to configure Kamailio in echo mode, i.e. SIP server repeats the input 
audio into the output, so I can hear myself with a very small amount of delay…

Is it possible? If so, how?

Thank you for your help.

Best regards,

From: Fomitchev, AndreiX
Sent: Thursday, March 29, 2012 1:34 PM
To: 'sr-users@lists.sip-router.org'
Subject: how to configure Kamailio in ECHO mode

Hi,

For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. 
SIP server
-
Intel Corporation SAS (French simplified joint stock company)
Registered headquarters: "Les Montalets"- 2, rue de Paris,
92196 Meudon Cedex, France
Registration Number:  302 456 199 R.C.S. NANTERRE
Capital: 4,572,000 Euros

This e-mail and any attachments may contain confidential material for
the sole use of the intended recipient(s). Any review or distribution
by others is strictly prohibited. If you are not the intended
recipient, please contact the sender and delete all copies.
-
Intel Corporation SAS (French simplified joint stock company)
Registered headquarters: "Les Montalets"- 2, rue de Paris, 
92196 Meudon Cedex, France
Registration Number:  302 456 199 R.C.S. NANTERRE
Capital: 4,572,000 Euros

This e-mail and any attachments may contain confidential material for
the sole use of the intended recipient(s). Any review or distribution
by others is strictly prohibited. If you are not the intended
recipient, please contact the sender and delete all copies.
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Re: [SR-Users] how to configure Kamailio in ECHO mode

[Alex Balashov]

Probably a user agent that is actually an endpoint, like a PBX or media server. 
 You should take a look at Asterisk or Freeswitch.

--
Alex Balashov - Principal 
Evariste Systems LLC 
235 E Ponce de Leon Ave 
Suite 106
Atlanta, GA 30030 
Tel: +1-678-954-0670 
Fax: +1-404-961-1892 
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

"Fomitchev, AndreiX"  wrote:

>Hi Alex,
>
>Thank you for your response. Please, could you tell me what I need to setup a 
>SIP “echo service”? I am new in SIP domain :/
>
>Regards,
>
>From: sr-users-boun...@lists.sip-router.org 
>[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Alex Balashov
>Sent: Thursday, March 29, 2012 1:38 PM
>To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users 
>Mailing List
>Subject: Re: [SR-Users] how to configure Kamailio in ECHO mode
>
>No, it's not possible. I think you have the wrong idea about what 
>Kamailio is. Kamailio neither receives nor produces audio at all.
>
>--
>Alex Balashov - Principal
>Evariste Systems LLC
>235 E Ponce de Leon Ave
>Suite 106
>Atlanta, GA 30030
>Tel: +1-678-954-0670
>Fax: +1-404-961-1892
>Web: http://www.evaristesys.com/, http://www.alexbalashov.com
>
>"Fomitchev, AndreiX" 
>mailto:andreix.fomitc...@intel.com>> wrote:
>Oups, sorry, bad manipulation:
>I want to configure Kamailio in echo mode, i.e. SIP server repeats the input 
>audio into the output, so I can hear myself with a very small amount of delay…
>
>Is it possible? If so, how?
>
>Thank you for your help.
>
>Best regards,
>
>From: Fomitchev, AndreiX
>Sent: Thursday, March 29, 2012 1:34 PM
>To: 'sr-users@lists.sip-router.org'
>Subject: how to configure Kamailio in ECHO mode
>
>Hi,
>
>For testing purpose, I want to configure a Kamailio server in ECHO mode, i.e. 
>SIP server
>-
>Intel Corporation SAS (French simplified joint stock company)
>Registered headquarters: "Les Montalets"- 2, rue de Paris,
>92196 Meudon Cedex, France
>Registration Number:  302 456 199 R.C.S. NANTERRE
>Capital: 4,572,000 Euros
>
>This e-mail and any attachments may contain confidential material for
>the sole use of the intended recipient(s). Any review or distribution
>by others is strictly prohibited. If you are not the intended
>recipient, please contact the sender and delete all copies.
>-
>Intel Corporation SAS (French simplified joint stock company)
>Registered headquarters: "Les Montalets"- 2, rue de Paris, 
>92196 Meudon Cedex, France
>Registration Number:  302 456 199 R.C.S. NANTERRE
>Capital: 4,572,000 Euros
>
>This e-mail and any attachments may contain confidential material for
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>recipient, please contact the sender and delete all copies.
>
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Re: [SR-Users] how to configure Kamailio in ECHO mode

[Daniel-Constantin Mierla]

On 3/29/12 1:48 PM, Alex Balashov wrote:
Probably a user agent that is actually an endpoint, like a PBX or 
media server. You should take a look at Asterisk or Freeswitch.


For a light weight media server like for audio echo purposes, SEMS (SIP 
Express Media Server) can be another option.


Cheers,
Daniel



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"Fomitchev, AndreiX"  wrote:

Hi Alex,

Thank you for your response. Please, could you tell me what I need to 
setup a SIP "echo service"? I am new in SIP domain :/


Regards,

*From:*sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] *On Behalf Of *Alex 
Balashov

*Sent:* Thursday, March 29, 2012 1:38 PM
*To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - 
Users Mailing List

*Subject:* Re: [SR-Users] how to configure Kamailio in ECHO mode

No, it's not possible. I think you have the wrong idea about what 
Kamailio is. Kamailio neither receives nor produces audio at all.


--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

"Fomitchev, AndreiX" > wrote:


Oups, sorry, bad manipulation:

I want to configure Kamailio in echo mode, i.e. SIP server repeats the 
input audio into the output, so I can hear myself with a very small 
amount of delay...


Is it possible? If so, how?

Thank you for your help.

Best regards,

*From:*Fomitchev, AndreiX
*Sent:* Thursday, March 29, 2012 1:34 PM
*To:* 'sr-users@lists.sip-router.org'
*Subject:* how to configure Kamailio in ECHO mode

Hi,

For testing purpose, I want to configure a Kamailio server in ECHO 
mode, i.e. SIP server


-
Intel Corporation SAS (French simplified joint stock company)
Registered headquarters: "Les Montalets"- 2, rue de Paris,
92196 Meudon Cedex, France
Registration Number:  302 456 199 R.C.S. NANTERRE
Capital: 4,572,000 Euros

This e-mail and any attachments may contain confidential material for
the sole use of the intended recipient(s). Any review or distribution
by others is strictly prohibited. If you are not the intended
recipient, please contact the sender and delete all copies.

-
Intel Corporation SAS (French simplified joint stock company)
Registered headquarters: "Les Montalets"- 2, rue de Paris,
92196 Meudon Cedex, France
Registration Number:  302 456 199 R.C.S. NANTERRE
Capital: 4,572,000 Euros

This e-mail and any attachments may contain confidential material for
the sole use of the intended recipient(s). Any review or distribution
by others is strictly prohibited. If you are not the intended
recipient, please contact the sender and delete all copies.


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Re: [SR-Users] Update of wiki write access

[Daniel-Constantin Mierla]

Thank you for feedback so far.

What is strength comparing with normal CAPTCHA?

As I understood (from http://www.recaptcha.net), one word (out of the 
two) in reCAPTCHA is not used for security, it can be typed wrong. So I 
wondered what is better in recaptcha than captcha...


Cheers,
Daniel

On 3/29/12 1:21 PM, Olle E. Johansson wrote:

29 mar 2012 kl. 09:45 skrev Daniel-Constantin Mierla:


Hello,

anyone has experiences with captcha vs recaptcha ( http://recaptcha.net/).

I'm using recaptcha on a couple of sites. It works fine for me.

On the downside, it dies with IPv6...
Hopefully google is going to fix that, at least to the World IPv6 Launch.
/O

There is a plugin for dokuwiki (http://www.dokuwiki.org/plugin:recaptcha) -- my 
interest is how strong the system is to defend against bot spammers, rather 
than other good faith things (i.e., helping scanning books for free).

Cheers,
Daniel

On 3/26/12 10:24 AM, Daniel-Constantin Mierla wrote:

Hello,

just to let you know that over the weekend the ACL (user authentication) for 
write access to wiki portals of the project has been turned on.

Anyone that wants to write to the wiki has to create herself/himself an 
account. The reason for requiring user authentication is weakness of the captha 
system used for anonymous posting -- during the past weeks lot of wiki pages 
were updated by spam bots.

If anyone is aware of a stronger captcha-like system to be used for anonymous 
posting, write back here. It has to work with dokuwiki.

Cheers,
Daniel


--
Daniel-Constantin Mierla
Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany

http://www.asipto.com/index.php/kamailio-advanced-training/
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Re: [SR-Users] Reload lua script?

[Daniel-Constantin Mierla]

Hello,

On 3/29/12 11:49 AM, Javier Gallart wrote:

Hello all

I'm starting to play with lua scripting in Kamailio. It's really cool, 
some complex stuff in the script becomes really easy this way. The 
script is becoming quite big, therefore I'd like to use lua_run 
instead of lua_dofile. My doubt is pretty  basic: if i load a script 
in memory at startup via the "load" parameter and the script is 
modifed afterwards, is there any way to reload the module so the 
script is reread?
the two contexts created by app_lua were designed one for loading at 
startup and the other for reloading every time. Reloading on demand at 
runtime is not possible. I thought of checking the timestamp of the 
script, but can be a bit risky if saving on intermediary state during 
updates. The scripts are loaded in each process, sending a mi/rpc 
command will require to do some inter-process signaling. Both doable, 
not in my short term plans, though, if someone is contributing, will be 
appreciated.


Some options for speed up with existing code:
- you can compile the Lua scripts, so reloading it each time will be 
faster (no more compile on-the-fly)
- not sure about this one: in your script loaded at startup you may have 
some logic to load another script based on your rules (e.g., file 
timestamp, periodically, ...)


Cheers,
Daniel

--
Daniel-Constantin Mierla
Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany
http://www.asipto.com/index.php/kamailio-advanced-training/


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Re: [SR-Users] early media: How to change to tag

[Iñaki Baz Castillo]

2012/3/29 Andreas Granig :
> I recently learned that for example Siemens switches implement "Request
> Disposition: no-fork" defined in http://www.ietf.org/rfc/rfc3841.txt,

Annoying spec. Now the client can tell the proxy "please don't fork"... WTF?

-- 
Iñaki Baz Castillo


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[SR-Users] branches usage

[Uri Shacked]

Thanks.
so, when i do append_branch and t_relay... i will send two invites at once,
right?
BR,
Uri
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Re: [SR-Users] how to configure Kamailio in ECHO mode

[Juha Heinanen]

Fomitchev, AndreiX writes:

> Thank you for your response. Please, could you tell me what I need to
> setup a SIP “echo service”? I am new in SIP domain :/

sems (iptel.org/sems) has echo plugin.

-- juha

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Re: [SR-Users] branches usage

[Carsten Bock]

Hi,

correct.

Carsten

2012/3/29 Uri Shacked :
> Thanks.
> so, when i do append_branch and t_relay... i will send two invites at once,
> right?
> BR,
> Uri



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[SR-Users] SEMS IVR issue

[Uri Shacked]

Hi,

If anyone here is familiar with SEMS
I installed it today, and could not figure out how to load the IVR
application... there is no ivr.so file
any ideas? there is so little info about it on the web...

BR,
Uri
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Re: [SR-Users] Reload lua script?

[Javier Gallart]

Thanks Daniel

On Thu, Mar 29, 2012 at 2:49 PM, Daniel-Constantin Mierla  wrote:

> Hello,
>
>
> On 3/29/12 11:49 AM, Javier Gallart wrote:
>
>> Hello all
>>
>> I'm starting to play with lua scripting in Kamailio. It's really cool,
>> some complex stuff in the script becomes really easy this way. The script
>> is becoming quite big, therefore I'd like to use lua_run instead of
>> lua_dofile. My doubt is pretty  basic: if i load a script in memory at
>> startup via the "load" parameter and the script is modifed afterwards, is
>> there any way to reload the module so the script is reread?
>>
> the two contexts created by app_lua were designed one for loading at
> startup and the other for reloading every time. Reloading on demand at
> runtime is not possible. I thought of checking the timestamp of the script,
> but can be a bit risky if saving on intermediary state during updates. The
> scripts are loaded in each process, sending a mi/rpc command will require
> to do some inter-process signaling. Both doable, not in my short term
> plans, though, if someone is contributing, will be appreciated.
>
Understood, we may give it a try in the next weeks.

>
> Some options for speed up with existing code:
> - you can compile the Lua scripts, so reloading it each time will be
> faster (no more compile on-the-fly)
>
I agree it's the most logical approach.

> - not sure about this one: in your script loaded at startup you may have
> some logic to load another script based on your rules (e.g., file
> timestamp, periodically, ...)
>
Not sure either, will try.

What about using luajit?

Regards

Javi

>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> Kamailio Advanced Training, April 23-26, 2012, Berlin, Germany
> http://www.asipto.com/index.**php/kamailio-advanced-**training/
>
>
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Re: [SR-Users] SEMS IVR issue

[Andreas Granig]

Hey Uri,

There's a dedicated mailing list for SEMS users:
http://lists.iptel.org/mailman/listinfo/sems

Regarding the docs, there are comprehensive Readmes for the different
modules in the doc/ folder. For IVR, there doesn't seem to be a
shared-object, because it looks like a set of python plugins. Check
doc/Readme.ivr.txt for instructions (can't tell you more about it as
I've never used that particular one, but it should give you a hint on
how to proceed further).

Andreas

On 03/29/2012 02:56 PM, Uri Shacked wrote:
> Hi,
>  
> If anyone here is familiar with SEMS
> I installed it today, and could not figure out how to load the IVR
> application... there is no ivr.so file
> any ideas? there is so little info about it on the web...
>  
> BR,
> Uri
> 
> 
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Re: [SR-Users] Missing "Route" in ACK or BYE request

[Daniel-Constantin Mierla]

Hello,

On 3/29/12 11:17 AM, Hervé Cochet wrote:

Hello,

I used kamailio to handle SIP line from a provider using uacreg table.
Everything was working fine till they add a proxy which do not set (or 
not propagate) route header in ACK or BYE request.


So my routing logic is broken in the "in-dialog" transaction mode, 
since the loose_route return FALSE.


If I try to force a t_relay after the loose_route, my message may not 
be routing properly if for example my UAC was not connected with the 
standard 5060 port or use a different protocol like TCP (It works if 
both are in UDP and use 5060 port).
Is there a way to retrieve the record-route previously sets for this 
session in order to route this message properly ?


is the Record-Route set mirrored in the 200ok?

If the device is not supporting record-routing, then ACK/BYE should not 
get to your proxy. Can you post a full sip trace for such a call (from 
invite to bye)? It will help to understand how the message flow is and 
maybe we can help more.


As a generic hint for storing/retrieving data, look at htable module, or 
if you prefer to use database storage, sqlops may be an alternative. You 
can keep the values based on callid.






Another problem is that the dialog module do not match the BYE 
transaction, the "did" variable is missing since the route is not 
there but it should match the request with SIP matching 
(dlg_match_mode is set to 1).


For example my dialog parameters are:

dialog::  hash=1214:1340198900
/state:: 3
timestart:: 1333006717
timeout:: 80537132
callid:: 16260-ci-30dd2867-797784...@my.sip.provider
from_uri:: sip:0367023...@my.sip.provider;user=phone
from_tag:: 16260-GI-30dd2868-17bb342c3
caller_contact:: sip:13.12.14.20:5060
caller_cseq:: 815264875
caller_route_set:: 
caller_bind_addr:: udp:130.120.140.131:5060
to_uri:: sip:0974711672@13.12.14.17;user=phone
to_tag:: 1878467993
callee_contact:: sip:0974711672@96.57.249.78:1024
callee_cseq:: 815264875
callee_route_set::
callee_bind_addr:: udp:130.120.140.131:5060/

The BYE request:
/BYE sip:0974711672@96.57.249.78:1024 SIP/2.0
Call-ID: 16260-ci-30dd2867-797784...@my.sip.provider
CSeq: 815264876 BYE
From: "0033482531303" 
;tag=16260-GI-30dd2868-17bb342c3 


Max-Forwards: 28
Record-Route: 
To: ;tag=1878467993
Via: SIP/2.0/UDP 13.12.14.20:5060;branch=z9hG4bK-LNVP-1196a924-5f1f495a
Reason: q.850;cause=16
User-Agent: Cirpack/v4.42a (gw_sip)
Content-Length: 0/

Why does this BYE request is not matched by the dialog modules using 
SIP parameters (Call-ID, uri, tag seems correct) ?


Can you try to execute dlg_manage()? If Route header is missing, then 
there is nothing that triggers automatically dialog matching.


Cheers,
Daniel



Hervé


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Re: [SR-Users] Missing "Route" in ACK or BYE request

[Hervé Cochet]

Thanks Daniel,

On 03/29/2012 04:40 PM, Daniel-Constantin Mierla wrote:

  
Hello,
  
On 3/29/12 11:17 AM, Hervé Cochet wrote:
  

Hello,

I used kamailio to handle SIP line from a provider using uacreg table.
Everything was working fine till they add a proxy which do not set (or
not propagate) route header in ACK or BYE request.

So my routing logic is broken in the "in-dialog" transaction mode,
since the loose_route return FALSE.

If I try to force a t_relay after the loose_route, my message may not
be routing properly if for example my UAC was not connected with the
standard 5060 port or use a different protocol like TCP (It works if
both are in UDP and use 5060 port).
Is there a way to retrieve the record-route previously sets for this
session in order to route this message properly ?
  
  
is the Record-Route set mirrored in the 200ok?

Yes
 
If the device is not supporting record-routing, then ACK/BYE should not
get to your proxy. Can you post a full sip trace for such a call (from
invite to bye)? It will help to understand how the message flow is and
maybe we can help more.

Ok, I will send you a complete call flow in private.
 
As a generic hint for storing/retrieving data, look at htable module,
or if you prefer to use database storage, sqlops may be an alternative.
You can keep the values based on callid.
  
   


Another problem is that the dialog module do not match the BYE
transaction, the "did" variable is missing since the route is not there
but it should match the request with SIP matching (dlg_match_mode is
set to 1).

For example my dialog parameters are:

dialog::  hash=1214:1340198900
    state:: 3
    timestart:: 1333006717
    timeout:: 80537132
    callid:: 16260-ci-30dd2867-797784...@my.sip.provider
    from_uri:: sip:0367023...@my.sip.provider;user=phone
    from_tag:: 16260-GI-30dd2868-17bb342c3
    caller_contact:: sip:13.12.14.20:5060
    caller_cseq:: 815264875
    caller_route_set:: 
    caller_bind_addr:: udp:130.120.140.131:5060
    to_uri:: sip:0974711672@13.12.14.17;user=phone
    to_tag:: 1878467993
    callee_contact:: sip:0974711672@96.57.249.78:1024
    callee_cseq:: 815264875
    callee_route_set:: 
    callee_bind_addr:: udp:130.120.140.131:5060

The BYE request:
BYE sip:0974711672@96.57.249.78:1024
SIP/2.0 
Call-ID: 16260-ci-30dd2867-797784...@my.sip.provider

CSeq: 815264876 BYE 
From: "0033482531303" ;tag=16260-GI-30dd2868-17bb342c3


Max-Forwards: 28 
Record-Route:  
To: ;tag=1878467993


Via: SIP/2.0/UDP 13.12.14.20:5060;branch=z9hG4bK-LNVP-1196a924-5f1f495a

Reason: q.850;cause=16 
User-Agent: Cirpack/v4.42a (gw_sip) 
Content-Length: 0

Why does this BYE request is not matched by the dialog modules using
SIP parameters (Call-ID, uri, tag seems correct) ?
  
  
Can you try to execute dlg_manage()? If Route header is missing, then
there is nothing that triggers automatically dialog matching.

It works, thanks. I was not aware that route trigger dialog matching.
 
Cheers,
Daniel
   

Hervé



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-- 








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Re: [SR-Users] how to configure Kamailio in ECHO mode

[Fomitchev, AndreiX]

Hi,

Thank you for your responses.

I built sems 1.4.2 from scratch using default Makefile and when I launched it, 
I was unable to establish a communication because sems met network errors:
{log start}
(...)
 [#b6476b70/4110] [run, udp_trsp.cpp:186] INFO: Started SIP server UDP 
transport on 127.0.0.1:5080
 [#b6a7cb70/4104] [send, AmRtpPacket.cpp:212] ERROR: while sending RTP packet: 
Invalid argument
 [#b6a7cb70/4104] [compile_and_send, AmRtpStream.cpp:313] ERROR: while sending 
RTP packet: Invalid argument.
 [#b7486b70/4094] [send, transport.cpp:114] ERROR: sendto: Invalid argument
 [#b7486b70/4094] [send_request, trans_layer.cpp:899] ERROR: Error from 
transport layer: Invalid argument
{log end}
So, I am disappointed :/

I know that it is not SEMS maillist but perhaps someone met already this 
problem?

Thanks & regards,

-Original Message-
From: sr-users-boun...@lists.sip-router.org 
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Juha Heinanen
Sent: Thursday, March 29, 2012 2:54 PM
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users 
Mailing List
Subject: Re: [SR-Users] how to configure Kamailio in ECHO mode

Fomitchev, AndreiX writes:

> Thank you for your response. Please, could you tell me what I need to 
> setup a SIP “echo service”? I am new in SIP domain :/

sems (iptel.org/sems) has echo plugin.

-- juha

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Re: [SR-Users] siptrace error...

[Krishna Kurapati]

Daniel,

I finally had some time to put a log after it failed to convert from to
socket.

ERROR: siptrace [siptrace.c:1704]: there is not a valid number port 5060??k5
Mar 29 13:30:32 siptest /usr/sbin/kamailio[19848]: ERROR: siptrace
[siptrace.c:1514]: ERROR: bad FROM tcp:10.1.0.6:5060 or TO tcp:
75.62.17.45:63343

The passing parameter to the function look fine. Well formed and null
terminated. Inside it is failing.
Any ideas

Attaching the code segments around the lines


Code around line 1514:

   /* Convert proto:ip:port to sockaddress union SRC IP */
if (pipport2su(from->s, &from_su, &proto)==-1 || (pipport2su(to->s,
&to_su, &proto)==-1))
{
LOG(L_ERR, "ERROR: bad FROM %s or TO %s\n", from->s, to->s);
goto error;
}


Code around line 1704:

   else {
/*the address contains a port number*/
*p = '\0';
p++;
port_str.s = p;
port_str.len = strlen(p);
LM_DBG("the port string is %s\n", p);
if(str2int(&port_str, &port_no) != 0 ) {
LM_ERR("there is not a valid number port %s\n", p);
goto error;
}
*p = '\0';
}






On Wed, Mar 28, 2012 at 2:16 PM, Daniel-Constantin Mierla  wrote:

>  have you changed the log line to print the port? can you print the entire
> string, with proto and ip to see if there are spaces/invalid characters
> (print it between special chars, like [])?
>
> The line is inside the mirroring traffic via hep protocol, is what you
> want to get, right?
>
> Cheers,
> Daniel
>
>
> On 3/28/12 12:53 PM, Krishna Kurapati wrote:
>
> I tried to pull null terminator to see if that fixes the issue. That is
> why there is shift in the lines. It is same as line 1701.
>
>  Krish Kura
>
> On Wed, Mar 28, 2012 at 3:57 AM, Daniel-Constantin Mierla <
> mico...@gmail.com> wrote:
>
>>  Hello,
>>
>> are you working with a custom siptrace module? the line in siptrace.c
>> does not match. If you changed the sources, paste here the lines in
>> siptrace.c around 1705, 10 before and 10 after.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 3/26/12 3:23 PM, Krishna Kurapati wrote:
>>
>>  Hi,
>>
>>  I keep getting this error in the sip_trace module:
>>
>>  Mar 26 06:19:26 siptest /usr/sbin/kamailio[4279]: ERROR: siptrace
>> [siptrace.c:1705]: there is not a valid number port 5060?/??
>>
>>  Krish Kura
>>
>>
>>  ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing 
>> listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>> --
>> Daniel-Constantin Mierla
>> Kamailio Advanced Training, April 23-26, 2012, Berlin, 
>> Germanyhttp://www.asipto.com/index.php/kamailio-advanced-training/
>>
>>
>
> --
> Daniel-Constantin Mierla
> Kamailio Advanced Training, April 23-26, 2012, Berlin, 
> Germanyhttp://www.asipto.com/index.php/kamailio-advanced-training/
>
>
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