[SR-Users] add_uri_param with variables

[Patrick Wakano]

Hello list,
Hope you all doing well!

We've been attempting to add a URI parameter to implement the trunk group
(tgrp and trunk-context) but discovered the add_uri_param() function only
works with constant string we can't use a pseudovar to inform the value
to be added. Anyone knows why such limitation? Can it be changed?
It is possible to do the same thing via the subst_uri function, but it
would be so much more elegant to use the function...

Thank you,
Kind regards,
Patrick Wakano
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] outbound flow tokens and kamailio restart

[Pan Christensen]

Bounce.

Med vennlig hilsen
Pan B. Christensen

From: sr-users  On Behalf Of Pan 
Christensen
Sent: onsdag 13. juni 2018 09:19
To: Kamailio (SER) - Users Mailing List 
Subject: [SR-Users] outbound flow tokens and kamailio restart

Hello all.

I have created a WebRTC to SIP gateway. I implemented it using the outbound 
module. If I restart Kamailio during a call, subsequent messages fail to be 
routed.

{1 322 BYE 218565972_26748892@x.x.x.x} INFO: 
outbound [outbound_mod.c:261]: decode_flow_token(): flow-token failed validation
{1 322 BYE 218565972_26748892@x.x.x.x} INFO: 
rr [loose.c:519]: process_outbound(): failed to decode flow token
{1 322 BYE 218565972_26748892@x.x.x.x} INFO: 
rr [loose.c:794]: after_loose(): failed to process outbound flow-token

Is it possible to make the flow token survive a restart?

With kind regards
Pan B. Christensen
Developer

Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-mail: pan.christen...@phonect.no
Mobile: 41 88 88 00


 [cid:image007.png@01D3A0E8.376921D0] 
[facebook_2] [LinkedIn_logo_initials (1)] 


___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Using a var as timeout value in dlg_set_timeout_by_profile function

[jenus]

Ok thats clear thanks for chekking, does the dlg_set_timeout support 
$var?


Thanks,

Jan

Henning Westerholt schreef op 2018-06-14 16:17:
Am Donnerstag, 14. Juni 2018, 09:30:44 CEST schrieb 
je...@cyberchaos.nl:

I do not see any errors in the logs, it looks like it just sets the
timer to 0 and disconnects the call.


Hello Jan,

I just looked quickly in the code, it seems that the timeout value is 
not
interpreted as "pseudo-variable", therefore its not possible to use 
something

like $var in this method right now. It is just read as a numeric value.

dialog.c, line 1342:
if(dlg_set_timeout_by_profile((struct dlg_profile_table *) profile,
   _s, atoi(timeout_str)) != 0)

This can of course changed in the module with a change in the code.

Best regards,

Henning



___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

[Yuriy Gorlichenko]

You can watch at the kazoo project examples if you want to avoid rtp proxy

On Thu, Jun 14, 2018, 23:26 Daniel Tryba  wrote:

> On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
> > From the logs I see the jssip throw this error:
> >
> > "Failed to set remote offer sdp: Called with SDP without DTLS
> fingerprint."
> >
> > I would like to avoid RTPEngine, because from what I understand,
> FreeSwitch
> > can handle the media part.
>
> IIRC I got the same error in my tries to transcode/bridge SIP over TLS
> with SRTP to just plain old SIP with RTP. I haven't put any effort in it
> to get it working. You'll need to play around with rtpengine
> offer/answer, I based my test on
>
> https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
> but I blamed my failure on an old rtpengine :)
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

[David Villasmil]

Yeah, you need to set the correct offer, i did that a while ago, but i
can't remember how i did it.

Check out https://github.com/havfo/WEBRTC-to-SIP

Hope it help.

David

On Thu, Jun 14, 2018, 22:26 Daniel Tryba  wrote:

> On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
> > From the logs I see the jssip throw this error:
> >
> > "Failed to set remote offer sdp: Called with SDP without DTLS
> fingerprint."
> >
> > I would like to avoid RTPEngine, because from what I understand,
> FreeSwitch
> > can handle the media part.
>
> IIRC I got the same error in my tries to transcode/bridge SIP over TLS
> with SRTP to just plain old SIP with RTP. I haven't put any effort in it
> to get it working. You'll need to play around with rtpengine
> offer/answer, I based my test on
>
> https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
> but I blamed my failure on an old rtpengine :)
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

[Daniel Tryba]

On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
> From the logs I see the jssip throw this error:
> 
> "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
> 
> I would like to avoid RTPEngine, because from what I understand, FreeSwitch
> can handle the media part.

IIRC I got the same error in my tries to transcode/bridge SIP over TLS
with SRTP to just plain old SIP with RTP. I haven't put any effort in it
to get it working. You'll need to play around with rtpengine
offer/answer, I based my test on
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
but I blamed my failure on an old rtpengine :)


___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Kamailio + FreeSwitch + WebRTC

[Emanuel Gianico]

 Hi everybody, I followed this tutorial

https://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc#kamailio_33x_and_freeswitch_12x_for_media_services_and_sbc

And it works fantastic!

The next step was to add WebRTC support, so I added WebSockets module to
enable web clients to register on kamailio. It works flawlessly and
webphone clients register OK! (Followed this
http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket )

Now, when I call from a softphone (eyeBeam) to the web client (jssip) the
other party reach okay, rings okay, but when I pickup de call (from the web
client), the softphone goes directly to the VoiceMail.

>From the logs I see the jssip throw this error:

"Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."

I would like to avoid RTPEngine, because from what I understand, FreeSwitch
can handle the media part.

Can somebody please have the kindness to guide me on how to enable webrtc
between Kamailio and FreeSwitch? If somebody needs to see the
"kamailio.cfg" please let me know, and i would upload the file to a gist.

Cheers,
Emanuel.
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] can't assign values to $fU

[Karthik Srinivasan]

Daniel,

I appreciate the information.  Than you very much.

Karthik

On Thu, Jun 14, 2018 at 1:18 AM, Daniel-Constantin Mierla  wrote:

> Hello,
>
> you have to put this from the perspective of: changes to the SIP message
> (headers and body) are not immediately reflected. So even if you do a
> replace or subst operation, changes are not visible. If you do remove_hf()
> or append_hf(), it happens the same.
>
> The FAQ has an entry for it:
>
>   - https://www.kamailio.org/wiki/tutorials/faq/main#why_
> changes_made_to_headers_or
>
> $fu/U/d used to be read only, we made them r/w for convenience, as a
> variable based operation instead of uac_replace_from().
>
> The r-uri variables ($ru, $rU, $rd, ...) are not pointing to the SIP
> message buffer, there is a special field (and buffer) inside the internal
> structure of Kamailio, that's why changes to it are visible. The
> corresponding variables pointing to the SIP message buffer are $ou, $oU, ...
> Cheers,
> Daniel
>
> On 13.06.18 23:05, Karthik Srinivasan wrote:
>
> Henning,
>
> Thanks for the explanation.  This does clear it up for me.
>
> Do you happen to know if there is a list of pseudo vars that fall under
> the non special case?   (a list for those psedo vars where
> msg_apply_changes needs to be called for the update to be reflected while
> in routing file processing that is.)
>
> Thanks,
>
> Karthik
>
> On Wed, Jun 13, 2018 at 1:39 PM, Henning Westerholt 
> wrote:
>
>> Am Mittwoch, 13. Juni 2018, 20:28:13 CEST schrieb Alex Balashov:
>> > On Wed, Jun 13, 2018 at 01:26:07PM -0500, Karthik Srinivasan wrote:
>> > > Could you explain why we need to call this function when manipulating
>> $fU
>> > > ?
>> >
>> > Some PV manipulations work that way, others don't. :-) "Because
>> > Kamailio".
>>
>> Don't want to dig into to much technical details here..
>>
>> But to give a bit more context, the Kamailio architecture related to SIP
>> message processing is optimized to avoid re-parsing of the message during
>> configuration processing. This works with so called "lumps" which are
>> more or
>> less like a programming patch file (e.g. change, delete parts). This
>> lumps are
>> applied shortly before sending the message out or if you call
>> msg_apply_changes().
>>
>> Some parts of the SIP message are accessed directly, because they are
>> "more
>> important" (like the request URI) are handled specially, some like the
>> From
>> user are done like a normal SIP header part as described above.
>>
>> For a bit more details and to look into the details, have a look to the
>> dbg_sip_msg([log_level], [facility]) function in the debugger module.
>>
>> Best regards,
>>
>> Henning
>>
>>
>
>
> ___
> Kamailio (SER) - Users Mailing 
> Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
> Kamailio World Conference -- www.kamailioworld.com
>
>
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio as outbound proxy for PBX

[Wilkins, Steve]

Current Outbound Call:
WebRTC Client => Asterisk =>SIPDevice, here the SIPDevice communicates back 
through Asterisk.

Desired Outbound Call:
WebRTC Client => Asterisk (which has an outbound_proxy set in pjsip) 
=>Kamailio=>SIPDevice and back the same way?  
The end goal being that the SIPDevice never sees Asterisk.

I have attempted doing the Desired.  Asterisk is indeed sending the outbound 
calls to Kamailio, but Kamailio is not contacting the SIPDevice (number@fqdn).


Thank you,


-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Daniel 
Tryba
Sent: Thursday, June 14, 2018 11:40 AM
To: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio as outbound proxy for PBX

On Thu, Jun 14, 2018 at 10:56:51AM +, Wilkins, Steve wrote:
> If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio 
> actually make the SIP call?

Ehhh, yes. Why wouldn't that be possible?
 
> At some point I would like outbound calls to be controlled by Kamailio 
> so that the outside endpoints never communicate with the PBX.
> Currently a call goes through Kamailio to Asterisk and then Asterisk 
> communicates with the endpoint, and this is night ideal

I don't see what kind of scenario you want to implement and what your current 
setup is.


___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] t_save_lumps() and failure_route

[Julien Chavanton]

Hi, I have a scenario where I am using.

t_save_lumps(), lookup() and then I can choose to drop some branches in
branch routes.

However, it is possible that I endup without any branch left, in this case
t_relay() is returning and no failure route is called/created.
At this point I need to recover the original message

I believe it is not possible to restore/reset the message without passing
by failure route ?

Else, I could trigger a failure route using t_relay(), is there any
"handy/clean" way to trigger failure_route() ?

Regards
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Kamailio as outbound proxy for PBX

[Daniel Tryba]

On Thu, Jun 14, 2018 at 10:56:51AM +, Wilkins, Steve wrote:
> If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio 
> actually make the SIP call?

Ehhh, yes. Why wouldn't that be possible?
 
> At some point I would like outbound calls to be controlled by Kamailio
> so that the outside endpoints never communicate with the PBX.
> Currently a call goes through Kamailio to Asterisk and then Asterisk
> communicates with the endpoint, and this is night ideal

I don't see what kind of scenario you want to implement and what your
current setup is.


___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Memory leak in tm with push notifications

[Jurijs Ivolga]

Hi Henning,

Thank you a lot!

With kind regards,

Jurijs

On Thu, Jun 14, 2018 at 5:02 PM, Henning Westerholt  wrote:

> Am Donnerstag, 14. Juni 2018, 08:31:58 CEST schrieb Jurijs Ivolga:
> > Thank you a lot for your input.
> >
> > But I was asking if there is a point to create patch from this 2 commits
> > and apply to 4.4. Is it worth? Or there is no way to make this work
> > properly on 4.4? As I see, some part of code what is touched by this 2
> > commits differs quite a lot, so I'm bit afraid to create patch and apply
> it
> > to our production servers, especially if I don't have a clue what it
> > affects. :)
>
> Hello Juris,
>
> In my opinion there is indeed a risk that after applying the patch to 4.4
> you
> will run into other problems because the patch does not fit 100%. TM is
> one of
> the most complicated modules, I would not suggest to fiddle with it if you
> don't have a clue, as you mentioned. ;-) There is of course the
> possibility to
> get somebody else to port the patch for you.
>
> But as I already wrote - there are other important bugs which are fixed
> only
> in 5.0 and 5.1. We maintain only the last two stable release, as a project
> policy.
>
> So I would recommend that you update your production systems instead of
> trying
> to re-fit this individual patch into the older code base.
>
> Best regards,
>
> Henning
>
> --
> If you like the work that I do in Kamailio, please consider supporting me
> on
> Patreon: https://www.patreon.com/henningw
>
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Using a var as timeout value in dlg_set_timeout_by_profile function

[Henning Westerholt]

Am Donnerstag, 14. Juni 2018, 09:30:44 CEST schrieb je...@cyberchaos.nl:
> I do not see any errors in the logs, it looks like it just sets the
> timer to 0 and disconnects the call.

Hello Jan,

I just looked quickly in the code, it seems that the timeout value is not 
interpreted as "pseudo-variable", therefore its not possible to use something 
like $var in this method right now. It is just read as a numeric value.

dialog.c, line 1342:
if(dlg_set_timeout_by_profile((struct dlg_profile_table *) profile,
   _s, atoi(timeout_str)) != 0)

This can of course changed in the module with a change in the code.

Best regards,

Henning

-- 
If you like the work that I do in Kamailio, please consider supporting me on 
Patreon: https://www.patreon.com/henningw

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Memory leak in tm with push notifications

[Henning Westerholt]

Am Donnerstag, 14. Juni 2018, 08:31:58 CEST schrieb Jurijs Ivolga:
> Thank you a lot for your input.
> 
> But I was asking if there is a point to create patch from this 2 commits
> and apply to 4.4. Is it worth? Or there is no way to make this work
> properly on 4.4? As I see, some part of code what is touched by this 2
> commits differs quite a lot, so I'm bit afraid to create patch and apply it
> to our production servers, especially if I don't have a clue what it
> affects. :)

Hello Juris,

In my opinion there is indeed a risk that after applying the patch to 4.4 you 
will run into other problems because the patch does not fit 100%. TM is one of 
the most complicated modules, I would not suggest to fiddle with it if you 
don't have a clue, as you mentioned. ;-) There is of course the possibility to 
get somebody else to port the patch for you.

But as I already wrote - there are other important bugs which are fixed only 
in 5.0 and 5.1. We maintain only the last two stable release, as a project 
policy. 

So I would recommend that you update your production systems instead of trying 
to re-fit this individual patch into the older code base.

Best regards,

Henning

-- 
If you like the work that I do in Kamailio, please consider supporting me on 
Patreon: https://www.patreon.com/henningw

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Kamailio as outbound proxy for PBX

[Wilkins, Steve]

Good Morning All!

If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio 
actually make the SIP call?

At some point I would like outbound calls to be controlled by Kamailio so that 
the outside endpoints never communicate with the PBX.
Currently a call goes through Kamailio to Asterisk and then Asterisk 
communicates with the endpoint, and this is night ideal

Thanks ALL
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] integrate huawei HSS with kamailio

[Markus Monka]

Hi,

could you share some details, how you solved the problem?

Best Regards
 Markus

On Thu, Jun 14, 2018 at 11:30 AM, eyas barhouk  wrote:

> solved dears
> thanks
>
>
>
> Sent from my Samsung Galaxy smartphone.
>
>
>  Original message 
> From: eyas barhouk 
> Date: 14/06/2018 2:23 am (GMT+02:00)
> To: sr-users@lists.kamailio.org
> Subject: [SR-Users] integrate huawei HSS with kamailio
>
> hello dears
>
> i'm trying to integrate kamailio IMS with Huawei Hss instead of Fraunhofer
> HSS , so is there any one has an experiment with this or tested it before
> ???
> if yes, could you please guide me how to do that ? or where to start ?
>
> thanks in advance
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>


-- 
Dipl. Ing.
Markus Monka - mmo...@sipgate.de
Telefon: +49 (0)211-63 55 55-23
Telefax: +49 (0)211-63 55 55-22

sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] integrate huawei HSS with kamailio

[eyas barhouk]

solved dears
thanks



Sent from my Samsung Galaxy smartphone.


 Original message 
From: eyas barhouk 
Date: 14/06/2018 2:23 am (GMT+02:00)
To: sr-users@lists.kamailio.org
Subject: [SR-Users] integrate huawei HSS with kamailio

hello dears

i'm trying to integrate kamailio IMS with Huawei Hss instead of Fraunhofer HSS 
, so is there any one has an experiment with this or tested it before ???
if yes, could you please guide me how to do that ? or where to start ?

thanks in advance
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Forbidden - Private identity not found (Authorization: username)

[eyas barhouk]

solved in the same way dear Carsten
thank you for your kind help




Sent from my Samsung Galaxy smartphone.


 Original message 
From: Carsten Bock 
Date: 14/06/2018 12:23 pm (GMT+02:00)
To: eyas barhouk 
Cc: "Kamailio (SER) - Users Mailing List" 
Subject: Re: [SR-Users] Forbidden - Private identity not found (Authorization: 
username)

Hi,

you need to look into the data provisioned in the HSS.

You should have an IMPI as well as an associated IMPU like this:
"09876993998754@IMS1.NET1"

We've tested Kamailio with a whole bunch of HSS's (from NSN, E///,
ZTE, SummaNetworks, ...), so it's definitely not an issue on Kamailio.
Huawei should work straight away.

Thanks,
Carsten
--

Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Millerntorplatz 1
20359 Hamburg / Germany

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

Office +49 40 5247593-40
Fax +49 40 5247593-99

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/


2018-06-12 9:24 GMT+02:00 eyas barhouk :
> hi Carsten
>
> i appreciate your kind reply
>
>>what you've provisioned in the HSS? Are you using Fraunhofer's HSS?
> No,i am using huawei HSS with sip digest "SDA" IMPI authentication
>
>>Can you show me your trace
> you can find the needed traces in attached
>
> thanks in advance
>
> 
> From: sr-users  on behalf of Carsten
> Bock 
> Sent: Tuesday, June 12, 2018 4:22 AM
> To: Kamailio (SER) - Users Mailing List
> Subject: Re: [SR-Users] Forbidden - Private identity not found
> (Authorization: username)
>
> Hi,
>
> you need to check your provisioning in the HSS. Can you show me your
> trace and tell me, what you've provisioned in the HSS? Are you using
> Fraunhofer's HSS?
>
> Thanks,
> Carsten
> --
>
> Carsten Bock
> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Millerntorplatz 1
> 20359 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:cars...@ng-voice.com
>
> Office +49 40 5247593-40
> Fax +49 40 5247593-99
>
> 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/
>
>
> 2018-06-12 0:35 GMT+02:00 eyas barhouk :
>> Hello dears
>>
>> when i was trying to build an IMS platform by using kamailio with sip
>> digest
>> authentication scheme and ims_auth module, i couldn't register and the
>> result was as following :
>>
>> 1-the first register message passed to the s-cscf  then S-cscf
>> replaying with the message “401 unauthorized - challenging the ue”
>> 2-  the second register pass to the S-cscf with the authorization header
>> then the S-CSCF replying with "Forbidden - Private identity not found
>> (Authorization: username)"
>>
>> So based on that could any one tell me what is the reason of this error
>> and
>> how to fix it ??
>>
>> Thanks in advance
>>
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Forbidden - Private identity not found (Authorization: username)

[Carsten Bock]

Hi,

you need to look into the data provisioned in the HSS.

You should have an IMPI as well as an associated IMPU like this:
"09876993998754@IMS1.NET1"

We've tested Kamailio with a whole bunch of HSS's (from NSN, E///,
ZTE, SummaNetworks, ...), so it's definitely not an issue on Kamailio.
Huawei should work straight away.

Thanks,
Carsten
--

Carsten Bock
CEO (Geschäftsführer)

ng-voice GmbH
Millerntorplatz 1
20359 Hamburg / Germany

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

Office +49 40 5247593-40
Fax +49 40 5247593-99

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/


2018-06-12 9:24 GMT+02:00 eyas barhouk :
> hi Carsten
>
> i appreciate your kind reply
>
>>what you've provisioned in the HSS? Are you using Fraunhofer's HSS?
> No,i am using huawei HSS with sip digest "SDA" IMPI authentication
>
>>Can you show me your trace
> you can find the needed traces in attached
>
> thanks in advance
>
> 
> From: sr-users  on behalf of Carsten
> Bock 
> Sent: Tuesday, June 12, 2018 4:22 AM
> To: Kamailio (SER) - Users Mailing List
> Subject: Re: [SR-Users] Forbidden - Private identity not found
> (Authorization: username)
>
> Hi,
>
> you need to check your provisioning in the HSS. Can you show me your
> trace and tell me, what you've provisioned in the HSS? Are you using
> Fraunhofer's HSS?
>
> Thanks,
> Carsten
> --
>
> Carsten Bock
> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Millerntorplatz 1
> 20359 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:cars...@ng-voice.com
>
> Office +49 40 5247593-40
> Fax +49 40 5247593-99
>
> 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/
>
>
> 2018-06-12 0:35 GMT+02:00 eyas barhouk :
>> Hello dears
>>
>> when i was trying to build an IMS platform by using kamailio with sip
>> digest
>> authentication scheme and ims_auth module, i couldn't register and the
>> result was as following :
>>
>> 1-the first register message passed to the s-cscf  then S-cscf
>> replaying with the message “401 unauthorized - challenging the ue”
>> 2-  the second register pass to the S-cscf with the authorization header
>> then the S-CSCF replying with "Forbidden - Private identity not found
>> (Authorization: username)"
>>
>> So based on that could any one tell me what is the reason of this error
>> and
>> how to fix it ??
>>
>> Thanks in advance
>>
>>
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] Memory leak in tm with push notifications

[Jurijs Ivolga]

Hi Henning,

Thank you a lot for your input.

But I was asking if there is a point to create patch from this 2 commits
and apply to 4.4. Is it worth? Or there is no way to make this work
properly on 4.4? As I see, some part of code what is touched by this 2
commits differs quite a lot, so I'm bit afraid to create patch and apply it
to our production servers, especially if I don't have a clue what it
affects. :)

With kind regards,

Jurijs

On Wed, Jun 13, 2018 at 11:02 PM, Henning Westerholt 
wrote:

> Am Mittwoch, 13. Juni 2018, 09:51:45 CEST schrieb Jurijs Ivolga:
> > I think I have this issue and I'm using 4.4 and I can't use master for
> now.
> > I tried to cherry pick this 2 commits, but unfortunately it do not work.
> >
> > For example commit 5fe2a1a1c67b550431dcae3c98701073f7edd953 make
> changes in
> > function t_continue_helper, but 4.4 do not has such function, it has with
> > slightly different name - t_continue.
> >
> > Same commit add line 258 in src/modules/tm/t_suspend.c, but in 4.4 this
> > part of code is slightly different. There is no " t->flags &=
> > ~T_ASYNC_CONTINUE; " line in same if statement.
> >
> > There is no way to remove line 390 from same file, cause in 4.4 that part
> > of code differs quite a lot.
> >
> > With second patch 72f5eaeeef0239ebd16a2d645b83e83eb1a2b506 there was
> much
> > less problems, but still, there is big difference in part of code near
> line
> > 592 of this commit, but probably in 4.4 i just need to update line 527
> and
> > change "UNREF_FREE(new_cell); " to " UNREF_FREE(new_cell, 0);"
> >
> >
> > Is it a worth to try to cherry pick this 2 commits or there are too much
> > changes between 4.4 and Master and no way to make this work properly?
>
> Hello Juris,
>
> I don't know much details about your setup. But if you don't use a lot of
> custom code that needs to be touched before you can go to 5.1 then an
> update
> should be not difficult. There are also some other important fixes, some
> of
> them security relevant, that you miss as well if you stay on 4.4.
>
> Best regards,
>
> Henning
>
> --
> If you like the work that I do in Kamailio, please consider supporting me
> on
> Patreon: https://www.patreon.com/henningw
>
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Re: [SR-Users] can't assign values to $fU

[Daniel-Constantin Mierla]

Hello,

you have to put this from the perspective of: changes to the SIP message
(headers and body) are not immediately reflected. So even if you do a
replace or subst operation, changes are not visible. If you do
remove_hf() or append_hf(), it happens the same.

The FAQ has an entry for it:

  -
https://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers_or

$fu/U/d used to be read only, we made them r/w for convenience, as a
variable based operation instead of uac_replace_from().

The r-uri variables ($ru, $rU, $rd, ...) are not pointing to the SIP
message buffer, there is a special field (and buffer) inside the
internal structure of Kamailio, that's why changes to it are visible.
The corresponding variables pointing to the SIP message buffer are $ou,
$oU, ...

Cheers,
Daniel

On 13.06.18 23:05, Karthik Srinivasan wrote:
> Henning,
>
> Thanks for the explanation.  This does clear it up for me.
>
> Do you happen to know if there is a list of pseudo vars that fall
> under the non special case?   (a list for those psedo vars where
> msg_apply_changes needs to be called for the update to be reflected
> while in routing file processing that is.)
>
> Thanks,
>
> Karthik
>
> On Wed, Jun 13, 2018 at 1:39 PM, Henning Westerholt  > wrote:
>
> Am Mittwoch, 13. Juni 2018, 20:28:13 CEST schrieb Alex Balashov:
> > On Wed, Jun 13, 2018 at 01:26:07PM -0500, Karthik Srinivasan wrote:
> > > Could you explain why we need to call this function when
> manipulating $fU
> > > ?
> >
> > Some PV manipulations work that way, others don't. :-) "Because
> > Kamailio".
>
> Don't want to dig into to much technical details here..
>
> But to give a bit more context, the Kamailio architecture related
> to SIP
> message processing is optimized to avoid re-parsing of the message
> during
> configuration processing. This works with so called "lumps" which
> are more or
> less like a programming patch file (e.g. change, delete parts).
> This lumps are
> applied shortly before sending the message out or if you call
> msg_apply_changes().
>
> Some parts of the SIP message are accessed directly, because they
> are "more
> important" (like the request URI) are handled specially, some like
> the From
> user are done like a normal SIP header part as described above.
>
> For a bit more details and to look into the details, have a look
> to the
> dbg_sip_msg([log_level], [facility]) function in the debugger module.
>
> Best regards,
>
> Henning
>
>
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference -- www.kamailioworld.com

___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users