Re: [Kamailio-Users] poll: project name

[Carsten Bock]

Hi,

i also take 4.
I mistyped the name several times... the old name (though not available
anymore) was much better to remember and more clear what the project was
about...

Carsten

Am Montag, den 11.08.2008, 13:50 +0300 schrieb Daniel-Constantin Mierla:
 Hello everybody,
 
 some people expressed inconveniences regarding the new name, so we would 
 like to get your opinion about, therefore this is a poll that will get 
 us to a decision. Here is the question:
 
 Do you like the name Kamailio?
 1) Yes, I love it. Please keep it.
 2) It is OK.
 3) I do not mind the name at all
 4) I do not like it
 5) I hate the name and we should change it
 
 There is no need of explanations, as it is up to personal taste and 
 decision. Just vote by option.
 
 Cheers,
 Daniel
 
 PS. for those interested...
 
 As short history about renaming story. On the 28th of March the board 
 was informed about issues with SER part of project name. You can easily 
 find who has the trademarks. On the 7th of April the board was informed 
 about an agreement for the deadline as of September 30, 2008 to use old 
 name. From there on, all old releases, documents, etc. can be referred 
 by the old name, but nothing new. Other few discussions about what 
 should/could be kept/changed followed and on 12th of May started the 
 discussion about the new name with two main direction, something 
 similar, to remind about old project or just something completely new, 
 to be used as brand name. Couple of proposals and discussions regarding 
 possible names followed up to 16th of July when was the deadline and the 
 start of vote. Majority was reached on 17th but it was required to have 
 opinions from all members of the board, last came from Bogdan, on the 
 28th of July.
 
 All discussions about migration were on the board mailing list. No 
 change were done to openser.org configuration, just the new domain with 
 kamailio in it was pointed to same server. Every step done was reported 
 to board, following a schedule discussed there. Another discussion was 
 about making the renaming and 1.4.0 release in same day. Being 
 practically impossible to do all the work in one day, it was decided to 
 do first the announcement about renaming. That happened on 28th when all 
 agreed with the new name.
 
 As you can see, there was no 5 months of renaming duration. There was no 
 change in openser.org DNS configuration undertaken with/without the 
 knowledge of all board members. The new name domains were reserved when 
 majority was achieved, in any case; dns entries for new name were 
 pointed to old server shortly after booking the domains.
 


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[Kamailio-Users] \0 in SIP-Messsages - uac_replac_from - Lumps - etc...

[Carsten Bock]

Hi,

short question:
I've noticed, when the UAC-Module (from 1.3) replaces the From-Headerin
subsequential requests, the From-Header is filled up with zeros.
For most clients this is absolutely fine (tested with Snom's, AVM's,
Cisco's and many more). I've noticed, that some Zyxel devices however do
not like this and ignore the request in this case...
Is this somehow defined in the RFC? Can somebody give me hint on a
chapter or something? Or does it break the RFC?

Thanks in advance,
Carsten



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Re: [Kamailio-Users] \0 in SIP-Messsages - uac_replac_from - Lumps - etc...

[Carsten Bock]

Hi,

this only happens, when the replacement-From-Header is shorter than the
existing one. The number of \0 represents the number of characters
removed from the request.
Also, i forgot to mention one other, important issue: This happens only,
if we terminate the call using the dialog-module... in all other cases
this is fine.
It also seems, that this only occurs in the old 1.3 release, in 1.4 this
looks ok.

Carsten



Am Mittwoch, den 22.04.2009, 21:55 +0200 schrieb Daniel-Constantin
Mierla:
 Hello,
 
 On 04/22/2009 06:26 PM, Carsten Bock wrote:
  Hi,
 
  short question:
  I've noticed, when the UAC-Module (from 1.3) replaces the From-Headerin
  subsequential requests, the From-Header is filled up with zeros.

 this is strange. How many zeros -- is there a relation between original 
 size and number of zeros? Does it happen in all cases?
 
 Cheers,
 Daniel
 
  For most clients this is absolutely fine (tested with Snom's, AVM's,
  Cisco's and many more). I've noticed, that some Zyxel devices however do
  not like this and ignore the request in this case...
  Is this somehow defined in the RFC? Can somebody give me hint on a
  chapter or something? Or does it break the RFC?
 
  Thanks in advance,
  Carsten
 
 
 
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Re: [Kamailio-Users] \0 in SIP-Messsages - uac_replac_from - Lumps - etc...

[Carsten Bock]

Hi,

unfortunately the changes between the 1.3 and 1.4 Version only affect
documentation...
Since the differences between dialog/dlg_request_within.c are also not
worth mentioning, i believe, the error lies somewhere in tm-module or
even in the core of 1.3.

Carsten

Am Donnerstag, den 23.04.2009, 18:54 +0200 schrieb Daniel-Constantin
Mierla:
 Hello,
 
 have you tried to trace major changes between 1.3 and 1.4versions in 
 uac/from.c?
 
 Cheers,
 Daniel
 
 
 On 04/23/2009 11:55 AM, Carsten Bock wrote:
  Hi,
 
  this only happens, when the replacement-From-Header is shorter than the
  existing one. The number of \0 represents the number of characters
  removed from the request.
  Also, i forgot to mention one other, important issue: This happens only,
  if we terminate the call using the dialog-module... in all other cases
  this is fine.
  It also seems, that this only occurs in the old 1.3 release, in 1.4 this
  looks ok.
 
  Carsten
 
 
 
  Am Mittwoch, den 22.04.2009, 21:55 +0200 schrieb Daniel-Constantin
  Mierla:

  Hello,
 
  On 04/22/2009 06:26 PM, Carsten Bock wrote:
  
  Hi,
 
  short question:
  I've noticed, when the UAC-Module (from 1.3) replaces the From-Headerin
  subsequential requests, the From-Header is filled up with zeros.


  this is strange. How many zeros -- is there a relation between original 
  size and number of zeros? Does it happen in all cases?
 
  Cheers,
  Daniel
 
  
  For most clients this is absolutely fine (tested with Snom's, AVM's,
  Cisco's and many more). I've noticed, that some Zyxel devices however do
  not like this and ignore the request in this case...
  Is this somehow defined in the RFC? Can somebody give me hint on a
  chapter or something? Or does it break the RFC?
 
  Thanks in advance,
  Carsten
 
 
 
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Re: [Kamailio-Users] \0 in SIP-Messsages - uac_replac_from - Lumps - etc...

[Carsten Bock]

Hi,

i believe it was fixed with revision 3982:

- added support for local and remove display name in the dialogs
created by TM (new requests generated by TM). This is required when
buidling high level applications (like IPPBX) on top of OpenSER

For us, i think we will update those systems to a more recent version in
the near future...

Carsten

Am Mittwoch, den 29.04.2009, 16:41 +0200 schrieb Daniel-Constantin
Mierla:
 Hello,
 
 On 04/29/2009 03:03 PM, Carsten Bock wrote:
  Hi,
 
  unfortunately the changes between the 1.3 and 1.4 Version only affect
  documentation...
  Since the differences between dialog/dlg_request_within.c are also not
  worth mentioning, i believe, the error lies somewhere in tm-module or
  even in the core of 1.3.

 it is changed/restored using the data lump system. I will check although 
 I do not remember big changes in the past, but 1.3 is quite old by now, 
 so my memory can fail...
 
 Cheers,
 Daniel
 
  Carsten
 
  Am Donnerstag, den 23.04.2009, 18:54 +0200 schrieb Daniel-Constantin
  Mierla:

  Hello,
 
  have you tried to trace major changes between 1.3 and 1.4versions in 
  uac/from.c?
 
  Cheers,
  Daniel
 
 
  On 04/23/2009 11:55 AM, Carsten Bock wrote:
  
  Hi,
 
  this only happens, when the replacement-From-Header is shorter than the
  existing one. The number of \0 represents the number of characters
  removed from the request.
  Also, i forgot to mention one other, important issue: This happens only,
  if we terminate the call using the dialog-module... in all other cases
  this is fine.
  It also seems, that this only occurs in the old 1.3 release, in 1.4 this
  looks ok.
 
  Carsten
 
 
 
  Am Mittwoch, den 22.04.2009, 21:55 +0200 schrieb Daniel-Constantin
  Mierla:


  Hello,
 
  On 04/22/2009 06:26 PM, Carsten Bock wrote:
  
  
  Hi,
 
  short question:
  I've noticed, when the UAC-Module (from 1.3) replaces the From-Headerin
  subsequential requests, the From-Header is filled up with zeros.



  this is strange. How many zeros -- is there a relation between original 
  size and number of zeros? Does it happen in all cases?
 
  Cheers,
  Daniel
 
  
  
  For most clients this is absolutely fine (tested with Snom's, AVM's,
  Cisco's and many more). I've noticed, that some Zyxel devices however do
  not like this and ignore the request in this case...
  Is this somehow defined in the RFC? Can somebody give me hint on a
  chapter or something? Or does it break the RFC?
 
  Thanks in advance,
  Carsten
 
 
 
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Re: [Kamailio-Users] \0 in SIP-Messsages - uac_replac_from - Lumps - etc...

[Carsten Bock]

Hi Daniel,

The method send_bye (in modules/dialog/dlg_req_within.c) uses the
TM-API to send the request:
result = d_tmb.t_request_within
(met, /* method*/
str_hdr,  /* extra headers*/
NULL,  /* body*/
dialog_info,   /* dialog structure*/
bye_reply_cb,  /* callback function*/
(void*)cell);  /* callback parameter*/

So i thought, this might apply for this issue.

Carsten

Am Mittwoch, den 29.04.2009, 17:35 +0200 schrieb Daniel-Constantin
Mierla:
 Hello,
 
 On 04/29/2009 07:20 PM, Carsten Bock wrote:
  Hi,
 
  i believe it was fixed with revision 3982:
 
  - added support for local and remove display name in the dialogs
  created by TM (new requests generated by TM). This is required when
  buidling high level applications (like IPPBX) on top of OpenSER
 
  For us, i think we will update those systems to a more recent version in
  the near future...

 thanks for investigating. I will have a look these days. The message 
 refers to messages generated by tm, though.
 
 Daniel
 
  Carsten
 
  Am Mittwoch, den 29.04.2009, 16:41 +0200 schrieb Daniel-Constantin
  Mierla:

  Hello,
 
  On 04/29/2009 03:03 PM, Carsten Bock wrote:
  
  Hi,
 
  unfortunately the changes between the 1.3 and 1.4 Version only affect
  documentation...
  Since the differences between dialog/dlg_request_within.c are also not
  worth mentioning, i believe, the error lies somewhere in tm-module or
  even in the core of 1.3.


  it is changed/restored using the data lump system. I will check although 
  I do not remember big changes in the past, but 1.3 is quite old by now, 
  so my memory can fail...
 
  Cheers,
  Daniel
 
  
  Carsten
 
  Am Donnerstag, den 23.04.2009, 18:54 +0200 schrieb Daniel-Constantin
  Mierla:


  Hello,
 
  have you tried to trace major changes between 1.3 and 1.4versions in 
  uac/from.c?
 
  Cheers,
  Daniel
 
 
  On 04/23/2009 11:55 AM, Carsten Bock wrote:
  
  
  Hi,
 
  this only happens, when the replacement-From-Header is shorter than the
  existing one. The number of \0 represents the number of characters
  removed from the request.
  Also, i forgot to mention one other, important issue: This happens only,
  if we terminate the call using the dialog-module... in all other cases
  this is fine.
  It also seems, that this only occurs in the old 1.3 release, in 1.4 this
  looks ok.
 
  Carsten
 
 
 
  Am Mittwoch, den 22.04.2009, 21:55 +0200 schrieb Daniel-Constantin
  Mierla:



  Hello,
 
  On 04/22/2009 06:26 PM, Carsten Bock wrote:
  
  
  
  Hi,
 
  short question:
  I've noticed, when the UAC-Module (from 1.3) replaces the 
  From-Headerin
  subsequential requests, the From-Header is filled up with zeros.




  this is strange. How many zeros -- is there a relation between 
  original 
  size and number of zeros? Does it happen in all cases?
 
  Cheers,
  Daniel
 
  
  
  
  For most clients this is absolutely fine (tested with Snom's, AVM's,
  Cisco's and many more). I've noticed, that some Zyxel devices however 
  do
  not like this and ignore the request in this case...
  Is this somehow defined in the RFC? Can somebody give me hint on a
  chapter or something? Or does it break the RFC?
 
  Thanks in advance,
  Carsten
 
 
 
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Re: [Kamailio-Users] Converting rtpproxy recorded .rtp file to .wav file

[Carsten Bock]

Hi,

1)
Did you try to post your problem on the RTP-Proxy-Users' List?
http://lists.rtpproxy.org/mailman/listinfo/users
Probably, you might get more help there
2)
Did you try to extract the Audio with Wireshark? If Wireshark can play the
Audio (and i assume it is correctly implemented), then the recorded stream
as such is correct. Then you can check, if the bug lies in rtpbreak (or as a
next step: sox). Just to limit the possible sources of the problem...

Carsten

2010/1/14 Vikram Ragukumar vraguku...@signalogic.com

 Hello,

 An update,

 I tried using sox to convert the two .raw files into 2 mono channel wave
 files. The command line i used is below :
 sox -r 8k -b -c 1 -u rtp.0.0.raw rtp0.wav
 sox -r 8k -b -c 1 -u rtp.1.0.raw rtp1.wav

 When i listen to the .wav files, i hear speech but it is buried in a lot of
 noise. During blank periods (periods of no speech) there is a constant
 volume high pitched noise. Also during periods of speech, there seems to be
 bursts of noise in the background.

 The other engineer i work with and i, think that it is possibly because
 non-speech data is being interpreted as speech.

 What switch options should i change while invoking sox from the command
 line to get rid of the noise?

 Thanks and Regards,
 Vikram.



 Vikram Ragukumar wrote:

 Hello,

 I used Kamailio+rtpproxy to record a session and rtpproxy outputs the
 following files

 long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp,
 long_file_name.o.rtcp

 http://www.rtpproxy.org/wiki/RTPproxy/FAQ
  From the Rtpproxy FAQ above, i tried to extract the audio using rtpbreak
 and sox.

 rtpbreak -W -r long_file_name.a.rtp
 rtpbreak -W -r long_file_name.o.rtp

 The above commands generate rtp.0.0.raw, rtp.1.0.raw.

 Then when i run sox using
 sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm -s
 out.wav i get the following errors :

 sox: invalid option -- -
 sox: -c must be given a number

 Is there a switch/anything else that i am missing ?

 Thanks in advance,
 Regards,
 Vikram.






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Re: [SR-Users] New Diameter IMS Module : can we use it as SWx interface

[Carsten Bock]

Hi,

you can basically implement any Diameter Application (or Emulation) using
my Diameter-Server module. I use it for an HSS and I wanted to use it for
an PCRF Emulator emulating the Rx-Interface, if time allows

Thanks,
Carsten

2017-07-06 19:12 GMT+02:00 Igor Potjevlesch <igor.potjevle...@gmail.com>:

> Dear Kamailio users,
>
>
>
> Recently has been added to Kamailio two new modules : CDP and CDP_AVP
>
> There is not the TS 29.273 in it but I wanted to know if we can use it to
> emulate SWx interface? or add it with a non-hardcoded solution?
>
> I have seen the kamailio conference online and one session told about
> emulate IMS HSS with those two modules.
>
>
>
> Is someone tried to do it and want to share is experience with me ?
>
> Some DIAMETER functions in these modules are closed from the SWx.
>
>
>
> Best regards
>
>
>
> *Igor Potjevlesch*
>
>
>
>
> <https://www.avast.com/sig-email?utm_medium=email_source=link_campaign=sig-email_content=emailclient>
>  Garanti
> sans virus. www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email_source=link_campaign=sig-email_content=emailclient>
> <#m_-4523271578643981940_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
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Re: [SR-Users] Subst_body M Audio

[Carsten Bock]

Hi,

it seems you are triggering RTPProxy after the subst_body (hence the
port and the parameter a=nortpproxy in the SDP).

Try removing RTPProxy.

Thanks,
Carsten

2017-07-13 11:23 GMT+02:00 Nicolas Breuer <nicolas.bre...@belcenter.biz>:
>
>
> Hello,
>
>
>
> I’m trying to replace the M part in the SDP Body.
>
>
>
> Seems not working as expected ( I changed the “s” part without problems )
>
>
>
> Code :
>
> if ( subst_body('/(^m=.*)/m=testing of rewrite.\r/') ) { $var(s)=""; };
>
>
>
>
>
> Here’s the incoming body message
>
>
>
> c=IN IP4 217.112.180.235.
>
> t=0 0.
>
> m=audio 49152 RTP/AVP 8 9 18 101.
>
> a=rtpmap:8 PCMA/8000.
>
> a=rtpmap:9 G722/8000.
>
> a=rtpmap:18 G729/8000.
>
> a=fmtp:18 annexb=no.
>
> a=rtpmap:101 telephone-event/8000.
>
> a=fmtp:101 0-15.
>
>
>
> After subst. ( note the .10840 on the m line)
>
>
>
> c=IN IP4 217.112.180.10.
>
> t=0 0.
>
> m=testing of rewrite..10840
>
> a=rtpmap:8 PCMA/8000.
>
> a=rtpmap:9 G722/8000.
>
> a=rtpmap:101 telephone-event/8000.
>
> a=fmtp:101 0-15.
>
> a=nortpproxy:yes.
>
>
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Re: [SR-Users] kamailio LTE HSS

[Carsten Bock]

Hi,

there is no such functionality in Kamailio yet (and at least from my
side currently not on the roadmap).

You can do a lot with Diameter in Kamailio today, so it is possible to
implement a full LTE HSS on top of it (see my presentation at
KamailioWorld: https://www.youtube.com/watch?v=fVxqHpKymP8).
But I'm not aware of anyone started to build one.

Thanks,
Carsten

2017-07-04 10:12 GMT+02:00 Ibrahim Almahi <eng.ibrahi...@yahoo.com>:
> Dear Expert Fellows,
> Good Day,
>
> I'm wonder whether kamailio developed an open source based LTE HSS handles
> both IMS and LTE procedures "LTE Attach, Location Update, Authentication,
> ".
>
> If it's not yet developed, is there any plan for it in the near future.
>
> Thanks and have a nice day.
>
> Ibrahim Almahi
> 966594996362
>
>
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Re: [SR-Users] SCTP Support for Rx/Dx interfaces

[Carsten Bock]

Hi Fred,

can you point me to the Spec, where SCTP is mandatory? It was never
mandatory, when we connected Kamailio to Ericsson, NSN, ZTE, Huawei, ... ;-)

Currently, it is not possible to use SCTP on Diameter, it's TCP only (i've
heard about some hacks...). But that's definitely on the roadmap.

Thanks,
Carsten

Am 15. Mai 2017 um 10:58 schrieb Fred Schulz <fsch...@blackned.de>:

> Hi everybody,
>
>
>
> as I can read in the docs is the support for SCTP possible via Module just
> for SIP.
>
>
>
> Is there a possibility to add SCTP support to Rx/Dx interfaces at ICSCF?
> As read in the specs SCTP is mandatory for that interface…
>
>
>
> Thanks
>
>
>
> Mit freundlichen Grüßen
>
> fred schulz
>
>
>
>
>
> e-mail: fsch...@blackned.de
>
> tel.: +49 8331 9959-6-00
>
> fax: +49 8331 9959-6-09
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Re: [SR-Users] URGENT! stateless dispatcher BYE issue

[Carsten Bock]

Hi,

even if your dispatcher is stateless, you should use loose_route().

Thanks,
Carsten

2017-05-10 22:34 GMT+02:00 Satish Patel <satish@gmail.com>:
> We have 1 stateless dispatcher and 3 proxy behind that dispatcher,
> when Proxy send BYE message then dispatcher bouncing back that BYE  to
> Proxy and they playing ping pong.  I think i am missing something..
>
> How BYE get routed?  in dispatcher i am not using loose_route()
> function also should i need to use that but again we running
> dispatcher in stateless.
>
> Here i put SIP trace:  https://pastebin.com/vp9JCdNP
>
>
> Dispatcher config, Very simple..
>
> route{
>
> remove_hf("Route");
>
> if (method == "INVITE") {
> record_route();
> }
>
>
> if ( !ds_select_dst("1", "0") ) {
> xlog("L_ERR", "Unable to route \n");
> sl_send_reply("500","Unable to route");
> break;
> }
>
>forward(uri:host, uri:port);
>
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20359 Hamburg / Germany

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

Office +49 40 5247593-40
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Re: [SR-Users] Kamailio IMS NAT issue: PRACK/ACK RURI changing in P-CSCF

[Carsten Bock]

Hi,

There is a common scenario, where Kamailio needs to be in front of e.g
SEMS: PCRF Integration for QoS. You need to access the IP of the sender and
the SDP.
You would have to integrate that into SEMS, but if you do, you can actually
drop Kamailio as a Proxy-CSCf completely...

Thanks,
Carsten

Am 21.06.2017 09:48 schrieb "Mojtaba" :

> Is it not better to use SEMS as SBC in the front of your network? And
> all P/S/I/CSCF have the private IP address.
> Thanks
> --Mojtaba Esfandiari.S
>
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Re: [SR-Users] [Kamailio-Business] Astricon 2017

[Carsten Bock]

Hi,

just a note:
I give a talk on how Asterisk fits into an Kamailio IMS deployment: On
thursday at 10 AM, don't be late! :-)

I'll be there, too!

See you there,
Carsten



2017-09-25 10:04 GMT+02:00 Daniel-Constantin Mierla <mico...@gmail.com>:
> Hello,
>
> Kamailio will be present at Astricon 2017 having a stand in the expo
> area (Orlando, FL, during October 3-5 -- more details at www.astricon.net).
>
> Should you participate at the event and have flyers/other materials with
> products/solutions/services that use Kamailio or giveaways, you are
> welcome to join the booth or just drop them there, we are more than
> happy to promote and spread them. Also, you are very welcome to just
> stop by and have a chat.
>
> Besides the activity in the exhibition area, I and Fred Posner will have
> presentations about using Kamailio together with Asterisk or other VoIP
> applications. Be sure you do not miss them, the agenda is available at:
>
>   - http://www.asterisk.org/community/astricon-user-conference/schedule
>
> Expect a consistent group from Kamailio community, it's a great event to
> be at!
>
> Looking forward to meeting many of you next week in Orlando!
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - www.asipto.com
> Kamailio World Conference - www.kamailioworld.com
>
>
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20359 Hamburg / Germany

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

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

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Re: [SR-Users] telcosuite3-crm sql

[Carsten Bock]

Hi,

you find all the SQL files in /var/www/management and ./rest, as you
need the REST-API for the Management-Portal.

Thanks,
Carsten

2017-08-31 8:17 GMT+02:00 Kordován Szabolcs <kos...@gmail.com>:
> Hi,
>
> I would like to try the IMS according to the Kamailio World 2017: Kamailio
> For IMS And VoLTE presentation.
> Unfortunately I have problem with telcosuite, the sql structure is missing.
> Where can I find the whole create.sql?
>
> Thanks for help.
> Regards,
> Szabolcs
>
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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
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Re: [SR-Users] telcosuite3-crm sql

[Carsten Bock]

Hi,

the inital use is ad...@volte.guru with passwort "test1234".

The Algorithm is described here:
http://php.net/manual/en/function.password-hash.php

Thanks,
Carsten

2017-08-31 8:49 GMT+02:00 Kordován Szabolcs <kos...@gmail.com>:
> Hi Carsten,
>
> Thanks for your fast reply.
> Now my question is that what is the demo admin's password? Or what is the
> algorithm of the password generation (I see it is not a simple md5), I can
> replace it in database.
>
> Regards,
> Szabolcs
>
> 2017-08-31 8:39 GMT+02:00 Carsten Bock <cars...@ng-voice.com>:
>>
>> Hi,
>>
>> you find all the SQL files in /var/www/management and ./rest, as you
>> need the REST-API for the Management-Portal.
>>
>> Thanks,
>> Carsten
>>
>> 2017-08-31 8:17 GMT+02:00 Kordován Szabolcs <kos...@gmail.com>:
>> > Hi,
>> >
>> > I would like to try the IMS according to the Kamailio World 2017:
>> > Kamailio
>> > For IMS And VoLTE presentation.
>> > Unfortunately I have problem with telcosuite, the sql structure is
>> > missing.
>> > Where can I find the whole create.sql?
>> >
>> > Thanks for help.
>> > Regards,
>> > Szabolcs
>> >
>> > ___
>> > Kamailio (SER) - Users Mailing List
>> > sr-users@lists.kamailio.org
>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>> >
>>
>>
>>
>> --
>> 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/
>>
>> ___
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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
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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
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Re: [SR-Users] Problems with AAR over Rx Interface

[Carsten Bock]

 AVP: Media-Type(520) l=16 f=VM- vnd=TGPP val=CONTROL (4)
> AVP: Codec-Data(524) l=28 f=VM- vnd=TGPP val=downlink\noffer\n
> AVP: Codec-Data(524) l=27 f=VM- vnd=TGPP val=uplink\nanswer\n
> AVP: Flow-Status(511) l=16 f=VM- vnd=TGPP val=ENABLED (2)
> AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP
> val=CHARGING_CORRELATION_EXCHANGE (1)
> AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP
> val=INDICATION_OF_LOSS_OF_BEARER (2)
> AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP
> val=INDICATION_OF_RECOVERY_OF_BEARER (3)
> AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP
> val=INDICATION_OF_RELEASE_OF_BEARER (4)
> AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP
> val=INDICATION_OF_ESTABLISHMENT_OF_BEARER (now void) (5)
> AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP val=IP-CAN_CHANGE (6)
> AVP: Specific-Action(513) l=16 f=VM- vnd=TGPP
> val=ACCESS_NETWORK_INFO_REPORT (12)
> AVP: Framed-IP-Address(8) l=12 f=-M- val=172.25.152.195
> AVP: Authorization-Lifetime(291) l=12 f=-M- val=7200
> AVP: Auth-Grace-Period(276) l=12 f=-M- val=0
> AVP: Session-Timeout(27) l=12 f=-M- val=7200
>
>
>
>
>
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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

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Re: [SR-Users] VoLTE Capable LTE Handsets which support IMS uisng IPv4

[Carsten Bock]

Hi,

i've tested Kamailio IMS with the following handsets on IPv4 only:
- Samsung S4/S5/S6/S6 Edge
- IPhone
- OnePlus 3/3T/5
- Microsoft Lumia (various models)

I am currently on holiday. I think in our APN reply we do not provide
any IPv6 P-CSCF, even if the client requests this.

Thanks,
Carsten



2017-10-18 15:17 GMT-04:00 Ron <ron.kamai...@mcleodnet.com>:
> I would like to support VoLTE using an IPv4 IMS network rather than IPv6
> network.  The handsets that I have been trying insist on the IMS network
> support either IPv6 or IPv4v6 dual stack.  After giving them IPv4v6
> connectivity, I am finding that even though the handsets asked for both IPv4
> and IPv6 P-CSCFs, they only try the IPv6 proxy (if only the IPv4 proxy is
> provided, they will not try at all).
>
>
>
> I am wondering of anyone has worked with any LTE handsets which are willing
> to support IMS using IPv4 only.
>
>
>
> Thanks,
>
> Ron
>
>
>
>
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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
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Re: [SR-Users] Send UPDATE to inform changes in SDP

[Carsten Bock]

Hi,

likely, the best solution is here to use RTPEngine, so the IP/Port
stays identical for each request... i've tested it with various VoLTE
phones and various PCRF's.

Thanks,
Carsten
--

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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
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2018-04-26 15:59 GMT+02:00 Alfonso Pinto <alfonso.pi...@gmail.com>:
> Hi Alex,
>
> Thanks for your answer. I was thinking more or less the same, but it seems I
> oversimplified the example and was missing an important point.
> This is IMS, so in the middle there are a lot more in the middle. It seems
> that this behaviour is normal when precondition is set and the only way to
> avoid the network to release the call on A side is to send the update.
> I've seen that in Kamailio 5 a new function was included in TM:
> https://www.kamailio.org/docs/modules/5.1.x/modules/tm.html#tm.f.t_uac_send
> In the code it looks like it's trying to use the current dialog, but that
> gives me doubts about CSEQ handling on the other side.
>
> Anyway, I will try unless someone is faster replying saying it doesn't work
> :)
>
> Thanks,
> Alfonso.
>
> On Thu, Apr 26, 2018 at 2:37 PM, Alex Balashov <abalas...@evaristesys.com>
> wrote:
>>
>> I would instead redirect my focus to why A is dropping the call in this
>> situation. It shouldn't be doing that.
>>
>> Per the standards, the first SDP answer must be the final SDP answer
>> (absent an update or reinvite) *of that endpoint*. There's no rule saying
>> that must be true of the dialog as a whole.
>>
>> -- Alex
>>
>> --
>> Sent via mobile, please forgive typos and brevity.
>>
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Re: [SR-Users] Forbidden - Private identity not found (Authorization: username)

[Carsten Bock]

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
--

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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
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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
>
>
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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
--

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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:
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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
>>
>>
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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
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Re: [SR-Users] Does Kamailio support Cx interface Diameter over SCTP ?

[Carsten Bock]

Hi,

we are using SCTP in one of our commercial deployments, but we haven't
found the time yet, to port that part back to the official repo. This
may take some time, but sooner or later we will add this to the
open-source Kamailio.

Thanks,
Carsten



2018-01-15 13:43 GMT+01:00 Tsvetan Filev <tsvetan.fi...@inno-networks.com>:
> Hi all.
>
> This question has already been asked in the past:
> https://lists.kamailio.org/pipermail/sr-users/2015-July/089083.html
> But I don't see any answer if this is supported.
>
> Can you tell me if it is possible to enable the IMS to talk to HSS using Cx
> (diameter) over SCTP ?
>
> In the source code (kamailio/src/modules/cdp/configparser.c) I can see only
> these parameters for Peer node:
>
> - FQDN
> - Realm
> - port
> - src_addr
>
> but there is no protocol parameter.
>
> I tried with the example VM here:
> https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/
> I started all services and dumped the traffic with wireshark. The transport
> is TCP.
> This VM uses an older version of kamailio:
> 
> kamailio -v
> version: kamailio 4.3.4 (x86_64/linux) 583dc0
> flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS,
> DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
> DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
> USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> id: 583dc0
> compiled on 21:01:04 Dec  1 2015 with gcc 4.9.2
> 
>
> So I installed the latest stable but it seems it also uses TCP.
>
> Regards.
>
>
> ___
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-- 
Carsten Bock
CEO (Geschäftsführer)

Meet us at MWC 2018 in Barcelona and see IMS and VoLTE in action!
(February 26th - March 1st, Hall 7 - Booth 7L80)

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:
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Re: [SR-Users] Coredump while exchanging diameter capabilities

[Carsten Bock]

S_HOOKS, DISABLE_NAGLE,
> USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC,
> TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
> USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST,
> HAVE_RESOLV_RES
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
> id: unknown
> compiled on 05:30:36 Jan 15 2018 with gcc 4.8.5
> ===
>
> OS is openSUSE Leap 42.3.
> Kernel: Linux linux-o12d 4.4.104-39-default #1 SMP Thu Jan 4 08:11:03 UTC
> 2018 (7db1912) x86_64 x86_64 x86_64 GNU/Linux
>
> Do you have any idea what might be wrong ?
> It could be bad config but still it should say something in the log without
> a crash.
>
> Regards.
>
>
> ___
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>



-- 
Carsten Bock
CEO (Geschäftsführer)

Meet us at MWC 2018 in Barcelona and see IMS and VoLTE in action!
(February 26th - March 1st, Hall 7 - Booth 7L80)

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/

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Re: [SR-Users] Coredump while exchanging diameter capabilities

[Carsten Bock]

Hi,

The log clearly states, what went wrong and where. There was insufficient
memory allocated at some point.
I'm traveling right now, but I will check either tonight from the Hotel or
tomorrow.

Thanks for reporting,
Carsten

Am 17.01.2018 09:51 schrieb "Tsvetan Filev" :

> Hi Rick.
>
> Yes I have coredumps and backtraces but I installed the rpms from here
> https://www.kamailio.org/wiki/packages/rpms (http://download.opensuse.org/
> repositories/home:/kamailio:/v5.1.x-rpms/openSUSE_Leap_42.3/ ) so I'm
> missing the debug symbols.
> Can I find the kamailio debug rpm (kamailio-debuginfo-5.1.0-21.7.x86_64)
> somewhere ?
>
> Once I have the symbols I will create an issue on GitHub.
>
> Here is what I get for now:
>
> # gdb kamailio core.kamailio.479.e6ada9851c9848a98237890cfddfaf
> e3.6560.151617742200
>
> GNU gdb (GDB; openSUSE Leap 42.3) 8.0.1
> Copyright (C) 2017 Free Software Foundation, Inc.
> License GPLv3+: GNU GPL version 3 or later  html> 
> This is free software: you are free to change and redistribute it.
> There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
> and "show warranty" for details.
> This GDB was configured as "x86_64-suse-linux".
> Type "show configuration" for configuration details.
> For bug reporting instructions, please see:
>  .
> Find the GDB manual and other documentation resources online at:
> 
> .
> For help, type "help".
> Type "apropos word" to search for commands related to "word"...
> Reading symbols from kamailio...(no debugging symbols found)...done.
> [New LWP 6560]
> [Thread debugging using libthread_db enabled]
> Using host libthread_db library "/lib64/libthread_db.so.1".
> Core was generated by `/usr/sbin/kamailio -DD -P
> /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamai'.
> Program terminated with signal SIGABRT, Aborted.
> #0  0x7f637dfb38c7 in __GI_raise (sig=sig@entry=6) at
> ../sysdeps/unix/sysv/linux/raise.c:55
> 55  ../sysdeps/unix/sysv/linux/raise.c: No such file or directory.
> Missing separate debuginfos, use: zypper install
> kamailio-debuginfo-5.1.0-21.7.x86_64
> (gdb) bt full
> #0  0x7f637dfb38c7 in __GI_raise (sig=sig@entry=6) at
> ../sysdeps/unix/sysv/linux/raise.c:55
> resultvar = 0
> pid = 6560
> selftid = 6560
> #1  0x7f637dfb4c9a in __GI_abort () at abort.c:78
> save_stage = 1
> act = {__sigaction_handler = {sa_handler = 0x28af0100,
> sa_sigaction = 0x28af0100}, sa_mask = {__val = {140724448160064, 0,
> 140065300084096, 6721723, 140065147406864, 140065144188928,
> 140065137572536, 140065144419712, 21474836483, 140065144188928,
>   9019276028016, 4, 0, 234976, 4, 8177296}}, sa_flags =
> 2123556608, sa_restorer = 0x0}
> sigs = {__val = {32, 0 }}
> #2  0x0066906e in qm_debug_frag ()
> No symbol table info available.
> #3  0x0066beaf in qm_free ()
> No symbol table info available.
> #4  0x006755f5 in qm_shm_free ()
> No symbol table info available.
> #5  0x7f6374c38a1d in AAAFreeAVP () from /usr/lib64/kamailio/modules/
> cdp.so
> No symbol table info available.
> #6  0x7f6374c144ac in AAAFreeAVPList () from
> /usr/lib64/kamailio/modules/cdp.so
> No symbol table info available.
> #7  0x7f6374beb0f9 in save_peer_applications () from
> /usr/lib64/kamailio/modules/cdp.so
> No symbol table info available.
> #8  0x7f6374beb16d in Process_CEA () from /usr/lib64/kamailio/modules/
> cdp.so
> No symbol table info available.
> #9  0x7f6374be4a22 in sm_process () from /usr/lib64/kamailio/modules/
> cdp.so
> No symbol table info available.
> #10 0x7f6374bd15d6 in receive_message () from
> /usr/lib64/kamailio/modules/cdp.so
> No symbol table info available.
> #11 0x7f6374bc7032 in do_receive () from /usr/lib64/kamailio/modules/
> cdp.so
> No symbol table info available.
> #12 0x7f6374bcae19 in receive_loop () from /usr/lib64/kamailio/modules/
> cdp.so
> No symbol table info available.
> #13 0x7f6374bc4563 in receiver_process () from
> /usr/lib64/kamailio/modules/cdp.so
> No symbol table info available.
> #14 0x7f6374be0c8c in diameter_peer_start () from
> /usr/lib64/kamailio/modules/cdp.so
> No symbol table info available.
> #15 0x7f6374bd2e5f in cdp_child_init () from
> /usr/lib64/kamailio/modules/cdp.so
> No symbol table info available.
> #16 0x0054b705 in init_mod_child ()
> No symbol table info available.
> #17 0x0054b422 in init_mod_child ()
> No symbol table info available.
> #18 0x0054b422 in init_mod_child ()
> No symbol table info available.
> #19 0x0054b422 in init_mod_child ()
> No symbol table info available.
> #20 0x0054b422 in init_mod_child ()
> No symbol table info available.
> #21 

Re: [SR-Users] IMS (ng-voice vs github)

[Carsten Bock]

Hi,

There aren't too many differences:
I carefully review and test the changes several weeks before putting them
into my repository for deployment.
It also contains some more private changes, where I haven't found the time
yet (due to a busy schedule - we are currently deploying IMS & VoLTE on
four continents) to merge them back into the official repo.

Thanks,
Carsten

Am 04.02.2018 14:38 schrieb "Amar Tinawi" :

> Hello Dears,
> can you please tell me what are the differences between the github version
> of ims modules and the ng-voice modules introduced by Mr Carsten repo ?
>
>
> Thanks in advance
>
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>
>
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Re: [SR-Users] kamailio Rx_AAR qusetions

[Carsten Bock]

Hi,

the IMS_QOS Module has not been designed to be used twice on a single
session. As it was designed to work on a Proxy-CSCF, you should have
an originating and one terminating leg, which are two different
sessions on the same server (after going through the IMS-Core).

Thanks,
Carsten

2018-02-12 4:27 GMT+01:00 Lv, Victor (NSB - CN/Qingdao)
<victor...@nokia-sbell.com>:
> Hi All,
>
>
>
> I am trying to send two Rx_AAR after Kamailio receive 200 OK with SDP:
>
>
>
> The first one for termination part is worked well, but in MT_aar_reply route
> block, it’s not works.
>
> The one I marked with red color, is there anyone know how configure Kamailio
> to send two Rx_AAR after receive 200 OK SDP?
>
>
>
> Thanks a lot!!
>
> Victor
>
> +++
>
>
>
> if (t_check_status("200")){
>
> xlog("L_DBG", "IMS: Received 200 nside
> orig_initial_reply\n");
>
>
>
> $var(aarret) = Rx_AAR("MT_aar_reply", "term","",2);
>
> xlog("L_DBG", "AAR return code is $var(aarret)\n");
>
> }
>
> on MT_aar_reply block:
>
>
>
> route[MT_aar_reply]
>
> {
>
> #this is async so to know status we have to check the reply avp
>
> switch ($avp(s:aar_return_code)) {
>
> case 1:
>
> xlog("L_DBG", "Diameter: Term AAR success on media
> authorization\n");
>
> break;
>
> default:
>
> xlog("L_ERR", "IMS: AAR failed Term\n");
>
> xlog("L_ERR", "IMS: ttag: "+
> "$avp(TTAG_CUSTOM_AVP)");
>
> xlog("L_ERR", "IMS: ftag: "+
> "$avp(FTAG_CUSTOM_AVP)");
>
> xlog("L_ERR", "IMS: callid: "+
> "$avp(CALLID_CUSTOM_AVP)");
>
> #comment this if you want to allow even if Rx fails
>
>
> if(dlg_get("$avp(CALLID_CUSTOM_AVP)","$avp(FTAG_CUSTOM_AVP)","$avp(TTAG_CUSTOM_AVP)")){
>
> dlg_terminate("all", "Sorry no QoS
> available");
>
> exit;
>
> }
>
>   $var(aarret) = Rx_AAR("MO_aar_reply", "term","",2);
>
>   xlog("L_DBG", "AAR return code is $var(aarret)\n");
>
>
>
>
>
>
> _______
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>



-- 
Carsten Bock
CEO (Geschäftsführer)

Meet us at MWC 2018 in Barcelona and see IMS and VoLTE in action!
(February 26th - March 1st, Hall 7 - Booth 7L80)

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/

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[SR-Users] RTPengine - Transcoding example available?

[Carsten Bock]

Hi,

Quick (and simple) question:

Can someone provide a working RTPengine request, to enable Transcoding?

I know, this is possible, but I thought before looking into deep, how this
should be configured for the rtpengine_offer() command, could someone
provide an example?

Thanks,
Carsten
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Re: [SR-Users] IMS: maybe a bug in S-CSCF: should we create an issue?

[Carsten Bock]

ption_data
> |
>
> +-+-+-+---+--+--+--+--+--+
>
> | 196 | sip:catt...@vict-ims.net|   0 | 1 | NULL | NULL
> | NULL | NULL |  catt...@vict-ims.netsip:catt...@vict-ims.net
> sip:+4350255872...@vict-ims.net  tel:+4350255872178 |
>
> | 197 | sip:+4350255872...@vict-ims.net |   0 | 1 | NULL | NULL
> | NULL | NULL |  catt...@vict-ims.netsip:catt...@vict-ims.net
> sip:+4350255872...@vict-ims.net  tel:+4350255872178 |
>
> | 198 | tel:+4350255872178  |   0 | 1 | NULL | NULL
> | NULL | NULL |  catt...@vict-ims.netsip:catt...@vict-ims.net
> sip:+4350255872...@vict-ims.net  tel:+4350255872178 |
>
> +-+-+-+---+--+--+--+--+------+
>
> 3 rows in set (0.00 sec)
>
>
>
>
>
> Cfg file of (.24):
>
> kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf",
> "enable_debug_file", 0)
>
> kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf",
> "matching_mode", 0)
>
> kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf",
> "db_url", "cluster://cluster1")
>
> kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf",
> "db_url", DB_URL)
>
> kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf",
> "db_mode", 2)
>
> kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf",
> "timer_interval", 10)
>
> kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf",
> "timer_procs", 4)
>
>
>
> lg
>
> P
>
>
>
>
> The information contained in this e-mail message is privileged and
> confidential and is for the exclusive use of the addressee. The person who
> receives this message and who is not the addressee, one of his employees or
> an agent entitled to hand it over to the addressee, is informed that he may
> not use, disclose or reproduce the contents thereof, and is kindly asked to
> notify the sender and delete the e-mail immediately.
>
>
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CEO (Geschäftsführer)

Meet us at MWC 2018 in Barcelona and see IMS and VoLTE in action!
(February 26th - March 1st, Hall 7 - Booth 7L80)

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:
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Re: [SR-Users] TLS cipher suites

[Carsten Bock]

Hi,

it also depends on the version of OpenSSL, Kamailio was compiled against

I can confirm, that Kamailio supports Elliptic Curve Diffie Hellmann
(ECDHE), as I added support for it... ;-)

Thanks,
Carsten

2018-01-03 9:46 GMT+01:00 Karsten Horsmann <khorsm...@gmail.com>:

> Hello,
>
>
> There is an ssldump example on kamailio.org wiki to see the cipher suits.
>
> AFAIK it depends on your certificate/ca  and how you create it.
>
> I see this with an test self-signed certificate that I did with one cipher
> only.
>
> And of course you client need support for it.
>
> Am 02.01.2018 5:16 nachm. schrieb "Steve" <smh2...@zoho.com>:
>
>> I have a question about deploying TLSv1.2 with Kamailio 4.3.4-1 on a
>> Lubuntu 16.4.3 desktop environment. I changed the Kamailio default
>> *tls.cfg* file under the section [server:default] to “method=TLSv1.2”
>> and am using OpenSSL 1.0.2g  from the Lubuntu repository. All the
>> programs were loaded through the Synaptic Package Manager.
>>
>> My question is whether this version of Kamailio supports the cipher suite
>> ECDHE-RSA-AES256-GCM-SHA384. My version of OpenSSL lists it as an option,
>> but the highest strength cipher that the Kamailio 4.3.4 server seems to
>> accept is RSA-AES256-GCM-SHA384. My (limited) understanding is that ECDHE
>> is a better method of key exchange than RSA because it is ephemeral with
>> forward secrecy.
>>
>> I used Wireshark to look at the connection protocols for sip clients
>> Jitsi and Blink with the Kamailio server. Jitsi offers only four cipher
>> choices of what I understand are considered compromised security TLS
>> protocols and it connected with the RSA-AES128-CBC-SHA cipher. Blink offers
>> 65 cipher choices, starting with ECDHE-RSA-AES256-GCM-SHA384. My Kamailio
>> server accepted the 29th offering on the list, RSA-AES256-GCM-SHA384.
>> Unless I am missing something, Kamailio 4.3.4 doesn’t seem to support
>> ephemeral DH key exchanges. Is there some other TLS configuration file or
>> setting for Kamailio that can be changed to allow this?
>>
>>
>>
>> <https://www.avast.com/sig-email?utm_medium=email_source=link_campaign=sig-email_content=emailclient_term=icon>
>>  Virus-free.
>> www.avast.com
>> <https://www.avast.com/sig-email?utm_medium=email_source=link_campaign=sig-email_content=emailclient_term=link>
>> <#m_-8452394494004720091_m_5244919164888980266_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>>
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Re: [SR-Users] Kamailio with CGNAT

[Carsten Bock]

Hi,

it's seems not related to CGNAT, but more due to a misconfiguration on
the Proxy-CSCF. It should evaluate the "alias" in the R-URI in order
to send it to the proper address.

Thanks,
Carsten
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ng-voice GmbH
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http://www.ng-voice.com
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Am Di., 28. Aug. 2018 um 13:19 Uhr schrieb Amar Tinawi :
>
> Hi Joel
>
> the Via's and Record routes is due to IMS structure, but it doesn't affect 
> the concept as i test it with a normal sip Proxy
>
> Proxy IP : 94.252.181.93 : 5050
> RTP Engine : 94.252.181.49
> I-cscf : 10.192.129.134
> s-cscf :10.192.129.134
>
> Actually i'm using what Mr Carsten introduce for IMS
>
> On Mon, Aug 27, 2018 at 8:33 PM Joel Serrano  wrote:
>>
>> Hi Amar,
>>
>> I had a quick look and I see a lot of via/route headers, I will look
>> properly later today, can you give me some details on what IP is what
>> in those traces for when I check them later?
>>
>> Also, are you using a kamailio config based on vanilla config
>> (specially regarding NAT handling and Contact/SDP updates/rewrites)?
>>
>>
>>
>>
>>
>> On Sun, Aug 26, 2018 at 11:39 PM, Amar Tinawi  wrote:
>> > it's Kamailio for signaling with RTPengine for media (also the RTP engine
>> > has public IP)
>> >
>> > any call to reach the user behind the CGNAT will fail because of the 200 OK
>> > msg not reaching the Proxy somehow (i checked, the massage is generated in
>> > the client)
>> > and any call initiated by the CGNAT client succeeded
>> >
>> > would you please check the attached
>> > many thanks
>> >
>> > On Mon, Aug 27, 2018 at 6:00 AM Joel Serrano  wrote:
>> >>
>> >> It would definitely be useful to see sip traces of the different scenarios
>> >> to try and find what the problem is.
>> >>
>> >> Can you give some details of what your setup looks like? Is kamailio
>> >> acting as a signaling proxy only or is it also handling RTP with
>> >> rtpengine/rtpproxy?
>> >>
>> >> On Sun, Aug 26, 2018 at 13:30 Amar Tinawi  wrote:
>> >>>
>> >>> Thanks Joel For reply
>> >>>
>> >>> when any user (Not Nated or behind Normal NAT)  is trying to reach a
>> >>> client behind CGNAT, the INVITE delivered and the client is start to
>> >>> ringing, but when answering, the answer is not delivered to the proxy, so
>> >>> not delivered to the calling client result to the call started in the
>> >>> CGNATED client and still in establishing in the calling one.
>> >>>
>> >>> The call is established successfully when the CGNATED send the INVITE to
>> >>> Not Nated client and two way audio.
>> >>>
>> >>> CGNATED to CGNATED not working as well .
>> >>>
>> >>>
>> >>>
>> >>> On Sun, Aug 26, 2018 at 7:57 PM Joel Serrano  wrote:
>> >>>>
>> >>>> Also, if the client is not behind CGNAT do you get the Ringing from
>> >>>> called party?
>> >>>>
>> >>>> Is called party the same in both cases?
>> >>>>
>> >>>>
>> >>>>
>> >>>> On Sun, Aug 26, 2018 at 09:54 Joel Serrano  wrote:
>> >>>>>
>> >>>>> Can you send the INVITE that proxy sends to called party?
>> >>>>>
>> >>>>> On Sun, Aug 26, 2018 at 02:05 Amar Tinawi 
>> >>>>> wrote:
>> >>>>>>
>> >>>>>> Hello
>> >>>>>> Although my question it's not related to Kamailio directly, but i
>> >>>>>> think someone could help in this strange behavior
>> >>>>>>
>> >>>>>> in our tests, we have some clients under private networks, where the
>> >>>>>> provider of those networks implements CGNAT (NAT444) for it's 
>> >>>>>> customers to
>> >>>>>> reach internet
>> >>>>>> (Client Private IP) --->(Home

Re: [SR-Users] dlg.end_dlg on RTP timeout

[Carsten Bock]

Hi Richard,

this is awesome! Thanks!

One question:
Currently, the XML-RPC URL has to be set in the config, right? I
frequently have the following setup:
- 2x Kamailio
- 2x RTPEngine
(and each Kamailio is connected to both RTPEngines, for redundancy and
maintenance reasons)

If I configure the URL statically, RTPEngine will only try to
terminate the dialog on one of the Kamailios, not from the Kamailio
where it received the call. If I understand the source-code correctly,
you store the information, from where the dialog was received anyway,
right? (created_from_addr / created_from in struct call). Would it be
feasable, to create the URL dynamically based on the Proxy where it
came from?

Thanks,
Carsten


--

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CEO (Geschäftsführer)

ng-voice GmbH
Millerntorplatz 1
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Am Do., 6. Sep. 2018 um 11:16 Uhr schrieb Richard Fuchs :
>
> This is now supported as per 
> https://github.com/sipwise/rtpengine/commit/89084da8d820919b44a0244e16e6701822070a72
>
> Cheers
>
> On 2018-09-05 05:39, Daniel-Constantin Mierla wrote:
>
> There is the dlg.terminate_dlg rpc command that requires callid, from-tag and 
> to-tag as parameters:
>
>   * 
> https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dlg.r.terminate_dlg
>
> So it expects something like:
>
> 
> 
> dlg.terminate_dlg
> 
> _CALLID_VALUE__
> _FROM_TAG_VALUE__
> _TO_TAG_VALUE__
> 
> 
>
> I planned to make the from-tag and to-tag optional for quite some time, but 
> didn't get the time for it yet.
>
> Cheers,
> Daniel
>
> On 05.09.18 08:52, Richard Fuchs wrote:
>
> Yup that's exactly right.
>
> It would be fairly simple to implement an additional XMLRPC format if there's 
> a particular one that's more friendly towards Kamailio.
>
> Cheers
>
> On 2018-09-05 02:42, Daniel-Constantin Mierla wrote:
>
> Looking quickly at the readme of rtpengine application and digging a bit with 
> google, it is something like rtpengine has to be started with
>
> -b http(s)://myrpcserver.ip/path -x 1
>
> and then the xmlrpc request is going to be sent to that url, having a format 
> like:
>
> 
> 
> teardown
> 
> _CALLID_VALUE__
> 
> 
>
> Is it right? If yes, then I can try to make a sample config that could handle 
> it using xmlops, xhttp and jsonrpcs modules.
>
> Cheers,
> Daniel
>
>
> On 05.09.18 08:24, Richard Fuchs wrote:
>
> It does an XMLRPC callback. Currently there's two formats for it, one is a 
> sems sbc teardown request (using the from-tag), the other is a generic 
> "teardown" command using the call ID.
>
> Cheers
>
> On 2018-09-04 07:52, Daniel-Constantin Mierla wrote:
>
> Hello,
>
> what do you get from rtpengine on rtp timeout? An RPC call back or an http 
> request?
>
> Cheers,
> Daniel
>
>
> On 04.09.18 12:48, Igor Olhovskiy wrote:
>
> Hi all!
>
> Is there any way to end dialog in Kamailio on RTP Timeout event on RTPEngine?
>
> Or only look at logs/redis database with external tool?
>
> Regards, Igor
>
>
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>
> --
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> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio World Conference -- www.kamailioworld.com
> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
>
>
>
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>
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> www.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio World Conference -- www.kamailioworld.com
> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
>
>
>
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>
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> www.twitter.com/miconda -- ww

Re: [SR-Users] BIG NEWS: Launching NODAILIO - Arcticficial Intelligence (tm) SIP Server

[Carsten Bock]

Hi,

Is this the new 5G, Block-Chain powered, self driving vehicle, everybody's
been talking about?

Thanks,
Carsten

P.S.: Since the stock value likely just doubled: Do I have any
stock-options?

Alex Balashov  schrieb am So., 1. Apr. 2018,
18:36:

> Most important thing about Nodailio is that it has no security
> vulnerabilities.
>
> None whatsoever.
>
> -- Alex
>
> --
> Sent via mobile, please forgive typos and brevity.
>
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Re: [SR-Users] Russian meddling in Kamailio project

[Carsten Bock]

And we all have just one goal: Keep Putin, the great protector of free will
and democracy, in power!
My Russian wife is working hard on a new module called "app_assembler"
which will allow writing the config in native assembler code. Migrating our
5g Config, which will directly copy every call to FSB, to "app_perl" was
simply just not obfuscating enough.

Thanks,
Carsten

Alex Balashov  schrieb am So., 1. Apr. 2018,
06:43:

> (Filed by the Geostrategic Open Source Alliance.)
>
> ATLANTA, GA (1 April 2018)--In a twist of events that has shocked the
> global
> technical community, the well-known open-source VoIP professional Alex
> Balashov
> has revealed himself to be a deep cover intelligence operative of the GRU
> (Main Intelligence Directorate of the military of the Russian Federation)
> and
> the FSB (the Federal Security Bureau, the successor agency to the
> Soviet-era
> KGB).
>
> Balashov was reported to be seeking US political asylum and immunity from
> prosecution in return for an exposé of the ways in which the Russian
> foreign
> intelligence apparatus has sought to influence the direction of the
> open-source
> communications project "Kamailio", widely used in telecommunications
> carrier,
> service provider and enterprise environments to deliver high-scalability
> routing services and other SIP (Session Initiation Protocol) platform
> building
> blocks.
>
> In connection with these sensitive ongoing negotiations, Balashov was
> debriefed
> on behalf of the US Government by Fred Posner of The Palner Group, a
> counterintelligence think tank and security consulting firm based in
> Gainesville, Florida. This debriefing was coordinated with the German BND
> (Bundesnachrichtendienst) agency and other agencies representing security
> cooperation partners of the European Union and NATO.
>
> GOSA have been able to obtain exclusive excerpts from the portions of this
> interview not deemed top secret:
>
>
> POSNER: Please state your full name, age, place of birth, physical
> characteristics, serial number and any other relevant professional asset
> identification markers.
>
> BALASHOV: [redacted]
>
> POSNER: You have made the claim that Kamailio has been infiltrated by
> Russian
> spy agencies. Given that it's an open-source project, that's a bold and
> provocative claim. Where's the evidence?
>
> BALASHOV: Infiltration is perhaps strong word from spy novels. Reality is
> less
> thrilling. As you are knowing from controversy about election of our
> Donald,
> covert global influence today is mainly question of soft power.
>
> POSNER: Is there an obvious way in which this applies to the Kamailio
> project
> you can show us? I remind you that you have staked your personal freedom on
> this issue.
>
> BALASHOV: Influence is delicate matter. Da, of course there are some
> superficial indications...
>
> POSNER: Such as?
>
> BALASHOV: For example module to support language Squirrel. What is
> Squirrel?
> Who uses it? If you take a look at code is a bit complicated, da? What
> does it
> do?
>
> POSNER: It is a bit complicated, But I'm not convinced. Sell me.
>
> BALASHOV: It was in fact committed to GitHub by former Eastern Bloc
> personality, yes? Very complicated code for unknown mystery language
> support
> from Eastern Europe...
>
> POSNER: Yes, from the main developer and leader of the project.
>
> BALASHOV: Hmm.
>
> POSNER: What else?
>
> BALASHOV: There is a Ukranian [redacted] who is in charge of many SIP
> captures
> with system which diverts them to central database, yes? Something with
> name
> from ancient Greek troubadour or myths maybe?
>
> POSNER: We'll have to look into that.
>
> BALASHOV: Kamailio World is held every year in East Berlin. Every year
> back to
> Berlin.
>
> POSNER: So what?
>
> BALASHOV: Where do most guests of it spend their night?
>
> POSNER: [audible crack of pistol whip] This is not a quiz show! I ask the
> questions here.
>
> BALASHOV: Yes, yes, okay, they are at the Park Inn at Alexanderplatz.
>
> POSNER: And?
>
> BALASHOV: The history of this hotel in East German times and Stasi presence
> there...
>
> POSNER: That aspect of history is well-known.
>
> BALASHOV: Da... then is clear.
>
> POSNER: What are the strategic goals of Russia with regard to open source
> communication infrastructure?
>
> BALASHOV: From news lately can be seen that is soft power leveraging and
> economic sabotage.
>
> POSNER: Economic sabotage? How will they achieve that with open source?
>
> BALASHOV: Again is question of delicate influence in small ways. This is
> not
> time of Arab dignitaries visiting KGB hotels in Moscow in 1970s or this
> kind of
> naked and obvious trick.
>
> POSNER: [raises pistol] You are here to provide specifics.
>
> BALASHOV: Yes yes, okay. Well, I am not policy architect at Khoroshovskoe
> Shosse, but general point of view with colleagues is that best approach to
> Western countries is to encourage kind of "boondoggles" [air quotes] which
> 

Re: [SR-Users] Select last URI from Record-Route

[Carsten Bock]

Hi,

Check for $route_uri, we use it to determine the session-case (no or Mt) on
the Proxy-CSCF.

Thanks,
Carsten

Am 16.03.2018 17:07 schrieb "John Petrini" :

> The simple answer to that is it's a last ditch effort to look up an AOR
> record on a reply but it's part of other logic we use to establish direct
> media.
>
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Re: [SR-Users] Missing documentation

[Carsten Bock]

Hi,

can you open a ticket on the GitHub Tracker for this? Otherwise this might
get lost at some point... I don't have time to take care of it immediately
as I am travelling this and next week, more likely I would take care of it
later in November.

Thanks,
Carsten
--

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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
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Am Di., 16. Okt. 2018 um 09:51 Uhr schrieb Pan Christensen <
pan.christen...@phonect.no>:

> The documentation for this module seems to be missing everywhere:
>
>
>
> I’ve checked 4.0, 4.1, 4.2, 4.3, 4.4, 5.0 and 5.1 in addition to stable
> and devel.
>
>
>
> *Pan B. Christensen *
> Utvikler
>
> Phonect AS
>
> Mail: pan.christen...@phonect.no
> + 47 41 88 88 00
>
>  [image: mail_footer]
>
>
>
> *From:* sr-users  * On Behalf Of *Pan
> Christensen
> *Sent:* tirsdag 16. oktober 2018 09:43
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* Re: [SR-Users] Missing documentation
>
>
>
> Also missing from devel:
>
>
> Not Found
>
> The requested URL /docs/modules/devel/modules/ims_usrloc_scscf.html was
> not found on this server.
> --
> Apache/2.4.10 (Debian) Server at www.kamailio.org Port 443
>
>
>
>
>
> *Pan B. Christensen *
> Utvikler
>
> Phonect AS
>
> Mail: pan.christen...@phonect.no
> + 47 41 88 88 00
>
>  [image: mail_footer]
>
>
>
> *From:* sr-users  *On Behalf Of *Pan
> Christensen
> *Sent:* tirsdag 16. oktober 2018 09:14
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* [SR-Users] Missing documentation
>
>
>
> Hello.
>
>
>
> The documentation for the ims_usrloc_scscf module seems to be missing:
>
>
>
> *Not Found*
>
> The requested URL /docs/modules/stable/modules/ims_usrloc_scscf.html was
> not found on this server.
> --
>
> *Apache/2.4.10 (Debian) Server at www.kamailio.org
> <http://www.kamailio.org> Port 443*
>
>
>
>
>
>
>
> *Pan B. Christensen *
> Utvikler
>
> Phonect AS
>
> Mail: pan.christen...@phonect.no
> + 47 41 88 88 00
>
>  [image: mail_footer]
>
>
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Re: [SR-Users] Crash ims_diameter_server

[Carsten Bock]

Hi,

my colleague implemented a fix for this some months ago, but we haven't
pushed it upstream. I will talk to him to fix this in the official Repo
also.

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

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Am Do., 1. Nov. 2018 um 06:55 Uhr schrieb Iman Mohammadi <
iman.mohammadi.tele...@gmail.com>:

> Hi
> I tried it before,it must be list for our purpose,int or string doesnt
> meet our purpose
>
> thanks
>
> On Thu, Nov 1, 2018 at 9:00 AM Mojtaba  wrote:
>
>> Hi.
>> There are some reason for this issue.
>> The three nested json is not reason for crash ims_diameter.
>> Let try this senario.
>> Chenge type of the last indoor level of json from List to other type(like
>> int or ...).
>> Then check the lamailio is crashed or not?
>> With Regards.Mojtaba
>>
>>
>> On Wed, 31 Oct 2018 21:56 Iman Mohammadi, <
>> iman.mohammadi.tele...@gmail.com> wrote:
>>
>>> When the below format is sent from rest , kamailio crashes
>>>
>>> List[{
>>> List[{
>>> List[{
>>>  ]}
>>>  ]}
>>>  ]}
>>> (Json with 3 nested lists),
>>> With 2 lists it works properly,
>>> When diameter is translated to json with 3 lists by this module it also
>>> works properly,
>>> Json format is correct too.
>>>
>>> gdb:
>>>
>>> gdb) bt full
>>> #0  0x7f5e23c4a11e in diameterserver_add_avp (m=0x0, d=0x7f5e1b5c2240 
>>> "", len=12, avp_code=431, flags=64, vendorid=0, data_do=2,
>>> func=0x7f5e23c51d6c <__FUNCTION__.20148> "parselist") at 
>>> avp_helper.c:208
>>> avp = 0x7f5e1b5c3bd0
>>> __FUNCTION__ = "diameterserver_add_avp"
>>> #1  0x7f5e23c4d0c8 in parselist (response=0x0, list=0x7fff9f21a8d0, 
>>> item=0x11b67d0, level=2) at avp_helper.c:309
>>> i = 1
>>> flags = 64
>>> x = "p\250!\237"
>>> avp_list = {head = 0x0, tail = 0x0}
>>> avp_list_s = {s = 0x7f5e1b5c2240 "", len = 12}
>>> __FUNCTION__ = "parselist"
>>> #2  0x7f5e23c4cffc in parselist (response=0x7f5e1b5c3ab8, list=0x0, 
>>> item=0x11b6550, level=1) at avp_helper.c:304
>>> subitem = 0x11b67d0
>>> i = 0
>>> flags = 64
>>> x = "\000\000\000"
>>> avp_list = {head = 0x0, tail = 0x0}
>>> avp_list_s = {s = 0x7fff9f21a8f0 "@\251!\237\377\177", len = 
>>> 600070897}
>>> __FUNCTION__ = "parselist"
>>> #3  0x7f5e23c4db3a in addAVPsfromJSON (response=0x7f5e1b5c3ab8, 
>>> json=0x7f5e23e53950 ) at avp_helper.c:349
>>> subitem = 0x11b6550
>>> i = 4
>>> __FUNCTION__ = "addAVPsfromJSON"
>>> root = 0x11b4210
>>> #4  0x7f5e23c3fbdb in callback_cdp_request (request_in=0x7f5e1b5c19b0, 
>>> param=0x0) at ims_diameter_server.c:193
>>> fmsg = 0xab7840 <_faked_msg>
>>> backup_rt = 1
>>> ctx = {rec_lev = 0, run_flags = 0, last_retcode = 1, jmp_env = 
>>> {{__jmpbuf = {1, -5713032302318786866, 7971288, 7971288,
>>> 140042162528420, 0, -5713032302339758386, 
>>> 5712822384154044110}, __mask_was_saved = 0, __saved_mask = {__val = {
>>>   140735863172072, 127, 0, 140042162532340, 
>>> 140042389578761, 4222451713, 140042162532340, 140042162532340, 
>>> 140042162532340,
>>>   140042162532340, 140042162532358, 140042162532467, 
>>> 140042162532340, 140042162532467, 0, 0}
>>> response = 0x7f5e1b5c3ab8
>>> __FUNCTION__ = "callback_cdp_request"
>>> #5  0x7f5e24a705c0 in api_callback (p=0x7f5e1b598f50, 
>>> msg=0x7f5e1b5c19b0, ptr=0x0) at api_process.c:83
>>> t = 0x7f5e1b598f50
>>> auto_drop = 32767
>>> h = 0x7f5e1b5b3358
>>> ---Type  to continue, or q  to quit---
>>> x = {type = (unknown: 2669784000), handler = {requestHandler = 
>>> 0x7f5e23c3eed1 ,
>

Re: [SR-Users] Core dump on diameter CCR request

[Carsten Bock]

Hi,

Also, what version of Kamailio are you using?

Thanks,
Carsten



Am Mo., 12. Nov. 2018, 09:50 hat Mojtaba  geschrieben:

> Hello,
> We developed OCS based on these module in Kamailio: ims_charging,
> ims_ocs, ims_dimameter_server and worker process written in Q_HSS. We
> have not this issue.
> Are you sure ims_charging is connected to any ocs proccess like ims_ocs?
> Because CCR/CCA and other related message is transfered between them.
> With Regards.
> Mojtaba Esfandiari.S
>
> On Mon, Nov 12, 2018 at 11:48 AM Daniel-Constantin Mierla
>  wrote:
> >
> > Hello,
> >
> > get the full backtrace with gdb from the core file -- it should reveal
> > where the crash happened.
> >
> > Also, it might be good to open an issue in bug tracker from
> > github.com/kamailio/kamailio in order to track it properly.
> >
> > Cheers,
> > Daniel
> >
> >
> > On 11.11.18 12:52, Venkatesh Macha wrote:
> > > Hi all,
> > >
> > > I am trying to implement Diameter charging with Kamailio. I am
> following
> > > below tutorial
> > >
> https://www.kamailio.org/events/2016-KamailioWorld/Day2/15-Carsten.Bock-Diameter-Charging-With-Kamailio.pdf
> > >
> > > But whenever I try to do the Ro_CCR from the INVITE block, Kamailio is
> > > crashing and generating the core dump. I am new to diameter. I might be
> > > doing something wrong here.
> > >
> > > please someone help me here.
> > >
> > > Please find the *diameter.xml* and *Diameter_core_dump_full_debug.txt*
> file.
> > >
> > > diameter.xml
> > > 
> > > Diameter_core_dump_full_Debug.txt
> > > <
> http://sip-router.1086192.n5.nabble.com/file/t6374/Diameter_core_dump_full_Debug.txt
> >
> > >
> > >
> > > Here is my kamailio.cfg :
> > >
> > > loadmodule "cdp"
> > > loadmodule "cdp_avp"
> > > loadmodule "ims_charging"
> > > loadmodule "ims_dialog"
> > >
> > > modparam("cdp","config_file","/etc/kamailio/diameter.xml")
> > > modparam("cdp", "debug_heavy", 1)
> > > # - ims_diameter_ro params -
> > > #modparam("ims_charging", "single_ro_session_per_dialog", 1)
> > > modparam("ims_charging", "origin_host", "10.195.0.38");
> > > modparam("ims_charging", "origin_realm", "10.195.0.38");
> > > modparam("ims_charging", "destination_realm", "mydomain.com");
> > > modparam("ims_charging","interim_update_credits",30);
> > > modparam("ims_charging","timer_buffer",5);
> > > # -- ims_dialog params --
> > > modparam("ims_dialog", "dlg_flag", 8)
> > >
> > >
> > > While starting the kamailio, i am getting following errors:
> > > Nov 11 11:40:03 ip-10-195-0-38 /usr/local/sbin/kamailio[25653]: ERROR:
> cdp
> > > [peerstatemachine.c:675]: add_peer_application(): Too many
> applications for
> > > this peer (max 0), not adding Application 4:0.
> > >
> > > Before crashing kamailio showing following error:
> > >
> > > Nov 11 11:39:49 ip-10-195-0-38 /usr/local/sbin/kamailio[25571]: INFO:
> {1
> > > 18325 INVITE Nf6-dl2WeXF.LJhXotiRsuMhijsNx45o} 

Re: [SR-Users] IMS: SMS over IP - how to get the originator's tel address?

[Carsten Bock]

Hi Ron,

You can't change it on the Proxy-CSCF, but on the HSS. According to 3GPP
Specs, the first identity associated to an IMPI is the "default-identity",
which is typically used by the phone and asserted by the Proxy-CSCF.

Thanks,
Carsten

Am Mi., 21. Nov. 2018, 00:31 hat Ron McLeod 
geschrieben:

>
>
>
>
> I am working on a VoLTE/IMS scenario to transfer text messages as SMS over
> IP (+g.3gpp.smsip feature) between UE’s.  This is a high-level view of the
> message flows between the endpoints (not showing CSCFs):
>
>
>
> UE1 IP-SM-GW
>
> | MESSAGE (SUBMIT) --->|
>
> |<--- 202 ACCEPTED |
>
> |  |
>
>  |< MESSAGE (SUBMIT_REPORT) |
>
> | 200 OK ->|
>
> |
>
> UE2 |
>
> |  |
>
>  |<-- MESSAGE (DELIVER) |
>
> | 200 OK ->|
>
>
>
> The messages does get transferred from UE1 to UE2 successfully.  Since
> this is GSM-style SMS, the OA (originating address) and DA (destination
> address) are specified using telephone numbers, not sip addresses.  This is
> not a problem for the DA, since it is specified in the SUBMIT TPDU from the
> originating handset, but when the IP-SM-GW sends the DELIVER TPDU it needs
> to specify the OA so that the recipient knows who sent the SMS, and this
> was not provided in the MESSAGE with the SMS-SUBMIT.
>
>
>
> According to 3GPP TS 24.341 (http://www.qtc.jp/3GPP/Specs/24341-b20.pdf):
>
> "The IP-SM-GW will have to use an address of the SM-over-IP sender that
> the SC can process (i.e. an E.164 number).  This address will come from a
> tel URI in a P-Asserted-Identity header (as defined in RFC 3325 [13])
> placed in the SIP MESSAGE request by the P-CSCF or S-CSCF."
>
>
>
> Looking at the MESSAGE request sent from the S-CSCF to the IP-SM-GW, it
> looks like neither the P-CSCF nor the S-CSCF is providing the tel address
> in a P-Asserted-Identity header:
>
>
>
> MESSAGE tel:+3108012345680 SIP/2.0
>
> Route: ,
>  :6060;lr;s=1;h=0;d=0;a=7369703a30303130333031323334353637383940696d732e6d6e633030332e6d63633030312e336770706e6574776f726b2e6f7267>
>
> P-Served-User:  >;sescase=orig;regstate=reg
>
> f: ;tag=2703358633
>
> t: 
>
> CSeq: 555874977 MESSAGE
>
> i: 2703358625_2324045032@2001:470:eb88:b021:27f:9507:589d:c201
>
> Via: SIP/2.0/UDP
> [2001:470:EB88:150:0:0:0:22]:6060;branch=z9hG4bKb857.bd3345ba81ec10b1d11639d9c7414606.0;i=a
>
> Via: SIP/2.0/TCP
> [2001:470:EB88:150:5054:FF:FE44:BFC9];branch=z9hG4bKb857.be53cc1c583b9c5f9dcebf3306f2b7c1.0
>
> v: SIP/2.0/UDP
> [2001:470:eb88:b021:27f:9507:589d:c201]:8906;rport=8013;branch=z9hG4bK3552622488
>
> Max-Forwards: 68
>
> P-Access-Network-Info: 3GPP-E-UTRAN-FDD;
> utran-cell-id-3gpp=001030003001
>
> c: application/vnd.3gpp.sms
>
> Allow: MESSAGE
>
> d: no-fork
>
> User-Agent: Xiaomi_Redmi 3S_Android6.0.1_9
>
> l: 27
>
> P-Charging-Vector: icid-value=49565351114B0100;
> icid-generated-at=2001:470:EB88:150:5054:FF:FE44:BFC9
>
> P-Asserted-Identity: <
> sip:001030123456...@ims.mnc003.mcc001.3gppnetwork.org>
>
> P-Visited-Network-ID: core.ims1.test
>
> Message Body
>
> GSM A-I/F RP - RP-DATA (MS to Network)
>
> Message Type RP-DATA (MS to Network)
>
> RP-Message Reference
>
> RP-Message Reference: 0x05 (5)
>
> RP-Originator Address
>
> RP-Destination Address - (3108012345680)
>
> RP-User Data
>
> Length: 14
>
> TPDU (not displayed)
>
>  GSM SMS TPDU (GSM 03.40) SMS-SUBMIT
>
>0...  = TP-RP: TP Reply Path parameter is not set in this
> SMS SUBMIT/DELIVER
>
>.0..  = TP-UDHI: The TP UD field contains only the short
> message
>
>..1.  = TP-SRR: A status report is requested
>
>...0 0... = TP-VPF: TP-VP field not present (0)
>
> .0.. = TP-RD: Instruct SC to accept duplicates
>
> ..01 = TP-MTI: SMS-SUBMIT (1)
>
>TP-MR: 92
>
>TP-Destination-Address - (8000)
>
>TP-PID: 0
>
>TP-DCS: 0
>
>TP-User-Data-Length: (5) depends on Data-Coding-Scheme
>
>TP-User-Data
>
> SMS text: HELLO
>
>
>
> Is this something that in configurable at the P-CSCF or the S-CSCF, or
> would it require now development?
>
>
>
> Thanks,
>
> Ron
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
___
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Re: [SR-Users] IMS: SMS over IP - how to get the originator's tel address?

[Carsten Bock]

Hi Ron,

You should check the response to the "200 OK" to a "REGISTER". It should
contain one or more "P-Associated-Identities", with the first one being the
default one. If no associated identity matches, the Proxy-CSCF will add the
default one as P-Asserted-Identity.

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/


Am Mi., 21. Nov. 2018 um 21:01 Uhr schrieb Ron McLeod <
ron.kamai...@mcleodnet.com>:

> Carsten – thanks for the reply.
>
>
>
> I am not trying to change it – the *tel* identity (and other public
> identities) is already defined in the profile stored in the HSS; I want the
> S-CSCF to insert P-Preferred-Identity headers for the IMPU’s associated
> with the UE, which it already knows since it downloaded the profile
> directly from the HSS.
>
>
>
> The P-CSCF and the handset itself *may* know these public identities as
> well if it would have SUBSCRIBEd for registration events from the S-CSCF,
> but in practice, I do not see the handset providing anything other than the
> IMPI, and the Kamailio-based P-CSCF with my current configuration does not
> insert any P-Asserted-Identity headers.
>
>
>
> TS 24.341 does suggest that either the P-CSCF or the S-CSCF could insert
> the header(s), and in the examples provided in Annex B.5 of the same
> document show the additional P-Asserted-Identity header being added by the
> S-CSCF before forwarding the request to the IP-SM-GW:
>
>
>
> *UE to P-CSCF*
>
> MESSAGE sip:sc.home1.net SIP/2.0
>
> Via: SIP/2.0/UDP [::aaa:bbb:ccc:ddd]:1357;comp=sigcomp;
> branch=z9hG4bKnashds7
>
> Max-Forwards: 70
>
> Route: , <
> sip:o...@scscf1.home1.net;lr>
>
> P-Preferred-Identity: "John Doe" 
>
> From: ; tag=171828
>
> To: 
>
> Call-ID: cb03a0s09a2sdfglkj490333
>
> Cseq: 666 MESSAGE
>
> Content-Type: application/vnd.3gpp.sms
>
> Content-Length: (...)
>
>
>
> *P-CSCF to S-CSCF*
>
> MESSAGE sip:sc.home1.net SIP/2.0
>
> Via: SIP/2.0/UDP pcscf1.visited1.net;branch=z9hG4bK240f341, SIP/2.0/UDP
>
> [::aaa:bbb:ccc:ddd]:1357;comp=sigcomp; branch=z9hG4bKnashds7
>
> Max-Forwards: 69
>
> Route: 
>
> P-Asserted-Identity: "John Doe" 
>
> From: ; tag=171828
>
> To: 
>
> Call-ID: cb03a0s09a2sdfglkj490333
>
> Cseq: 666 MESSAGE
>
> Content-Type: application/vnd.3gpp.sms
>
> Content-Length: (...)
>
>
>
> *S-CSCF to IP-SM-GW*
>
> MESSAGE sip:sc.home1.net SIP/2.0
>
> Via: SIP/2.0/UDP scscf1.home1.net;branch=z9hG4bK344a651, SIP/2.0/UDP
>
> pcscf1.visited1.net;branch=z9hG4bK240f341, SIP/2.0/UDP
>
> [::aaa:bbb:ccc:ddd]:1357;comp=sigcomp; branch=z9hG4bKnashds7
>
> Max-Forwards: 68
>
> Route: , <
> sip:cb03a0s09a2sdfglkj490...@scscf1.home1.net;lr>
>
> P-Asserted-Identity: "John Doe" 
>
> P-Asserted-Identity: tel:+1212555
>
> From: ; tag=171828
>
> To: 
>
> Call-ID: cb03a0s09a2sdfglkj490333
>
> Cseq: 666 MESSAGE
>
> Content-Type: application/vnd.3gpp.sms
>
> Content-Length: (...)
>
>
>
> Ron
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.kamailio.org] *On Behalf
> Of *Carsten Bock
> *Sent:* Wednesday, November 21, 2018 12:35 AM
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] IMS: SMS over IP - how to get the originator's
> tel address?
>
>
>
> Hi Ron,
>
>
>
> You can't change it on the Proxy-CSCF, but on the HSS. According to 3GPP
> Specs, the first identity associated to an IMPI is the "default-identity",
> which is typically used by the phone and asserted by the Proxy-CSCF.
>
>
>
> Thanks,
>
> Carsten
>
>
>
> Am Mi., 21. Nov. 2018, 00:31 hat Ron McLeod 
> geschrieben:
>
>
>
>
>
> I am working on a VoLTE/IMS scenario to transfer text messages as SMS over
> IP (+g.3gpp.smsip feature) between UE’s.  This is a high-level view of the
> message flows between the endpoints (not showing CSCFs):
>
>
>
> UE1 IP-SM-GW
>
> | MESSAGE (SUBMIT) --->|
>
> |<--- 202 ACCEPTED |
>
> |  |
>
>  |< MESSAGE (SUBMIT_REPORT) |
>
> | 200 OK ->|
>
&

Re: [SR-Users] SIP Express Media Server (aka SEMS)

[Carsten Bock]

Hi,

it depends on your configuration, e.g.:
- if you enable RTP-Relaying, then of course the flow for RTP would be UA1
<=> RTPEngine1 <=> SEMS <=> RTPEngine2 <=> UA2
- if you enable Transcoding (this forces RTP-Relaying to be active), it's
the same

If you don't use any of these options, then SEMS will do signalling only,
so the flow for RTP would be UA1 <=> RTPEngine1 <=> RTPEngine2 <=> UA2

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/


Am Mi., 9. Jan. 2019 um 12:36 Uhr schrieb Abhijit Dutta :

> Thanks Carsten and Mojtaba.
>
> I will need inputs/comments from you regarding the following scenarion.
>
> UA1   Kamailio/RTPEngine (Proxy1) --  SEMS
> -- Kamailio/RTPEngine (Proxy2) UA2
>
> If UA1 makes a video call (webrtc etc.)  how, the media will flow.
>
>1. Audio (RTP)  will traverse UA1 - RTPEngine1- SEMS - RTPEngine2-UA2
>2. Video (RTP) ? will it be via SEMS or that can be bypassed
>
> Thanks
> Abhijit
>
>
> --
> *From:* sr-users  on behalf of
> Carsten Bock 
> *Sent:* Wednesday, January 9, 2019 10:39 AM
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] SIP Express Media Server (aka SEMS)
>
> Hi,
>
> just another note:
> SEMS can forward Video-RTP Packets (but nothing beyond that).
>
> It's gotten a bit quiet on the SEMS-Project, but I don't think it's
> completely dead...
>
> 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/
>
>
> Am Mi., 9. Jan. 2019 um 11:30 Uhr schrieb Mojtaba :
>
> Hello,
> I've never heard that SEMS Support video, But I just know if you switch
> enable_rtprelay's option off, It would  works as p2p.
> So using these feature of SEMS is optional and you could keep using SEMS
> with no worry!
>
>
> On Wed, Jan 9, 2019 at 12:35 PM Abhijit Dutta  wrote:
>
> Hi All,
>
> Is the development work for SEMS server still on or stopped? Does it
> support video?
>
> Thanks
> Abhijit
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> --Mojtaba Esfandiari.S
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
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Re: [SR-Users] SIP Express Media Server (aka SEMS)

[Carsten Bock]

Hi,

just another note:
SEMS can forward Video-RTP Packets (but nothing beyond that).

It's gotten a bit quiet on the SEMS-Project, but I don't think it's
completely dead...

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/


Am Mi., 9. Jan. 2019 um 11:30 Uhr schrieb Mojtaba :

> Hello,
> I've never heard that SEMS Support video, But I just know if you switch
> enable_rtprelay's option off, It would  works as p2p.
> So using these feature of SEMS is optional and you could keep using SEMS
> with no worry!
>
>
> On Wed, Jan 9, 2019 at 12:35 PM Abhijit Dutta  wrote:
>
>> Hi All,
>>
>> Is the development work for SEMS server still on or stopped? Does it
>> support video?
>>
>> Thanks
>> Abhijit
>> ___
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
> --
> --Mojtaba Esfandiari.S
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] [sr-dev] RFC: updates to some core functions

[Carsten Bock]

Hi,

given the fact, that we do a lot of "non-sip" related stuff with Kamailio
(we are using Kamailio as a Diameter HSS, Diameter Charging-Server (OCS),
Diameter Routing-Agent (DRA)), I would also vote for number (2):

> 2) remove the function export from the core and export one with the
same
> name from the corex module

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/


Am Mi., 19. Dez. 2018 um 12:55 Uhr schrieb YAS0 CANER <
caner_y...@hotmail.com>:

> TmEXIT , kEXIT and corEXIT is failed 
> --
> *From:* sr-users  on behalf of Olle
> E. Johansson 
> *Sent:* Wednesday, December 19, 2018 2:38 PM
> *To:* Daniel-Constantin Mierla
> *Cc:* Kamailio (SER) - Development Mailing List; Kamailio (SER) - Users
> Mailing List
> *Subject:* Re: [SR-Users] [sr-dev] RFC: updates to some core functions
>
>
>
> > On 19 Dec 2018, at 12:26, Daniel-Constantin Mierla 
> wrote:
> >
> >
> > On 19.12.18 09:47, Olle E. Johansson wrote:
> >>
> >>> On 19 Dec 2018, at 09:41, Daniel-Constantin Mierla 
> wrote:
> >>>
> >>> corex module was added to hold the functions that otherwise would be
> >>> more or less "in the core", like those that were updated to support
> >>> variables in the parameters, so this is the one to take the place of
> >>> core in regard to exporting functions.
> >>>
> >>> tmx was added because tm module became very big, but also to try to
> >>> separate a bit between transaction management code and some functions
> in
> >>> top of it, in the way that tmx can work only with exported api by tm,
> so
> >>> if one adds a function there doesn't get access to all internals of
> >>> transaction and it is safer not to mess up things there. It is more or
> >>> less like usrloc and registrar, usrloc does internal management of
> >>> location records, registrar is the interface to configuration file (but
> >>> there are other modules on top of usrloc, like pua_usrloc, dmq_usrloc,
> ...).
> >>>
> >>> kex is the one that collected some functions that use to be in kamailio
> >>> (or better said openser at that time) but not in ser during 2005-2008
> >>> and can be a module that be analyzed to see if can be merged into other
> >>> modules. A big chunk of it used to be related to MI commands, but as we
> >>> got rid of MI, might be easier now to split parts of it and relocate.
> >>>
> >> Ok, so removing kex is a good first step for the coming release.
> >>
> >> It’s really hard explaining TMX and TM for new Kamailians.
> >
> > We can add a note at the top of docs for each of these modules to refer
> > to the other.
> >
> > On the other hand, I do not like to have a huge module. It is not
> > suitable for small embedded systems. Also, there are other modules using
> > the tm api, so it is a common approach. The tmx is exporting mostly
> > functions at higher level of transaction interaction, one can build a
> > transaction stateful sip routing without it, only using tm.
> >
> Damn it. My campaign for exit of the x modules just died. Sad story.
>
> Thanks for all the responses!
>
> /O
> > Cheers,
> > Daniel
> >
> >>
> >> /O :-)
> >>
> >>> Cheers,
> >>> Daniel
> >>>
> >>> On 19.12.18 09:25, Olle E. Johansson wrote:
> >>>> Going back one step, are there any reasons to keep tmx, kex and corex
> modules at all?
> >>>>
> >>>> At this point in the project I think many of the functions should be
> merged into
> >>>> the main modules and core.
> >>>>
> >>>> If I remember correctly, they exist because of a multi-brand history
> that is not
> >>>> really the case any more.
> >>>>
> >>>> /O
> >>>> “The campaign to remove Kamailio extensions to Kamailio”
> >>>>
> >>>>
> >>>>> On 19 Dec 2018, at 09:11, Henning Westerholt 
> wrote:
> >>>>>
> >>>>> Am Mittwoch, 19. Dezember 2018, 09:03:26 CET schrieb Sergey Safarov:

[SR-Users] Job Offering: Software Engineer & Dev-Ops for VoIP & IMS

[Carsten Bock]

Hi everyone,

due to our ongoing growth in the area of mobile network infrastructure, we
are currently looking into expanding our team. We are especially looking
for:

- Software Engineer VoIP & IMS (
https://www.ng-voice.com/software-engineer-voip-ims)
- Dev-Ops (VoIP & IMS) (https://www.ng-voice.com/dev-ops-voip-ims)

About us
ng-voice was founded by a team of leading open-source and VoIP pioneers and
passionate coders to develop software for mobile network operators. Our
virtualized, cloud native solutions allow entrepreneurial-thinking mobile
network operators to become software-centric and offer voice services on
any 4G, 5G, WiFi or any other data network – from regular mobile network
operators to private networks in remote areas. We are growing the team both
in Hamburg and Berlin and are looking for new colleagues to join us.

Your responsibilities
• Help us deploying and integrating our software solutions
• Have fun testing, validating and optimizing our solutions
• Enjoy building things and solving complex problems
• Proactively identify ways to unlock operational efficiency
• Take care of your teammates and have fun growing a company

Our requirements
We are more interested in your experience and knowledge than in formal
degrees:
• Strong interest in telephony
• Experience with signalling standards (e.g. SIP, DIAMETER)
• Fluent in English
• Efficient at driving deadlines and keeping yourself and others on topic

Our technology stack
• Docker; experience with Docker Swarm and Kubernetes is beneficial
• Kamailio, RTPEngine, FreeSwitch
• Redis, MySQL and RabbitMQ
• SIP, RTP, DIAMETER

What we offer
• Flexible working hours and great office locations (Berlin and Hamburg)
• Entrepreneurial culture and flat hierarchies
• High degree of autonomy and no dress code
• Competitive salary

Please send your application to j...@ng-voice.com. Let us know how your
profile fits our requirements. And please let us know the earliest date you
could join us plus your salary expectations.

We look forward to getting to know you!

Thanks,
Carsten
--
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Millerntorplatz 1 I 20359 Hamburg I Germany
www.ng-voice.com

Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49
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Re: [SR-Users] PRE-PAY update timer

[Carsten Bock]

Hi Steven,

If you want, I can provide you an example config based on ims_charging and
ims_ocs without IMS. Even though, we do not use ims_ocs anymore in new
deployments, we still have it running in production.

Thanks,
Carsten

Stephen Bucklin  schrieb am Sa., 25. Mai 2019,
01:31:

> Alex,
> Thanks so much for that. A lot to play with in the morning.
> I did set up an rtimer that called a route that then did checks if the
> number of dialogs > 1.
> I think I will keep playing with the time slices as the user may divert,
> conference etc so keeping the charging real time is my goal. There is also
> the potential of data usage at the same time. The credit is calculated on
> set up for the dialled path (mo and term costs) if each call and the
> charging calculator will then "reserve" money for 30 seconds and allow
> connection. Every 30 seconds it would request another 30 using a uuid
> passed in set up.
> I am hoping this will minimise Balance inconsistency for multiple calls +
> sms + data, with the actual accounting being done upon completion.
> I did not even think of jsonrpc, that will keep me reading tomorrow!
> Thanks,
> Steve
>
> On Sat, 25 May 2019 at 00:17, Alex Balashov 
> wrote:
>
>> Well, the generic answer to the question of how to periodically do
>> something in the background is 'rtimer':
>>
>> https://kamailio.org/docs/modules/5.2.x/modules/rtimer.html
>>
>> This can be combined with putting dialogs in an `htable` (perhaps using
>> the dialog module's start/end event_routes) and walking over the
>> entries in a background rtimer route:
>>
>>
>> https://kamailio.org/docs/modules/5.2.x/modules/htable.html#htable.f.sht_iterator_start
>>
>> Alternately, one can use jsonrpc_exec()
>>
>>
>> https://kamailio.org/docs/modules/5.2.x/modules/jsonrpcs.html#jsonrpcs.f.jsonrpc_exec
>>
>> to run 'dlg.list':
>>
>> https://kamailio.org/docs/modules/5.2.x/modules/dialog.html#dlg.r.list
>>
>> ... and iterate over the output with the help of the 'jansson' module
>> (JSON parser):
>>
>> https://kamailio.org/docs/modules/5.2.x/modules/jansson.html
>>
>>
>> But honestly, I'd ask yourself whether this extra layer of complexity is
>> truly worth it. For a lot of prepaid setups, live balance decrementing
>> isn't worth the bother.
>>
>> An alternate and simpler approach is to just let the call be initiated
>> if the balance is > 0 at the time the call is made, then deduct from the
>> balance when the call ends.
>>
>> Yeah, it means someone's free to make a 4 hour call without the prepaid
>> credit to support it, and worse yet, possibly a number of 4 hour calls
>> initiated simultaneously. But most calls aren't like that, and from a
>> business perspective, is it really worth the headache of live
>> decrementing and all the moving parts involved?
>>
>> Maybe it is, but I would at least ask the question.
>>
>> -- Alex
>>
>> On Fri, May 24, 2019 at 06:18:19PM +0100, Steve Bucklin wrote:
>>
>> > Hello all (again),
>> >
>> > I am looking at a 'good' method to start a call with "chunks" of time
>> > allocated. I wish the initial call to check a credit, and if credit is
>> good,
>> > allow a small amount of time (say 30 seconds). I have enabled DIALOG and
>> > have looked at CNXCC, but am confused!
>> >
>> > When the initial 30 seconds runs out, I want to authorise another 30
>> > seconds, etc, etc, etc. I have started to play and set a dialog timer
>> that
>> > expires to a route, and then can 'return' allowing the call to continue
>> -
>> > BUT was not able to reset the dialog timer to trip again after another
>> 30
>> > seconds.
>> >
>> > I tried to 'set_max_time' on cnxcc, but this did not seem to trip the
>> event
>> > route upon timeout?
>> >
>> > Does anyone have a pointer to any examples? I also am unsure of how to
>> > update the timer and keep a call in play!
>> >
>> > Steve
>> >
>> >
>> >
>> > ___
>> > Kamailio (SER) - Users Mailing List
>> > sr-users@lists.kamailio.org
>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>> --
>> Alex Balashov | Principal | Evariste Systems LLC
>>
>> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>
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>>
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Re: [SR-Users] IMS - change the transport layer protocol for Diameter based interfaces (especially Rx)

[Carsten Bock]

Hi Björn,

this misunderstanding was my fault. We have a drop-in replacement for the
Diameter-Stack developed in house (based on FreeDiameter), which we
originally wanted to publish in the Kamailio project. This is what I was
referring to in that talk.

However, after discovering several "commercial" IMS solutions based on
Kamailio (without mentioning Kamailio), some bad client discussions ("Why
should we pay for something, if we get the same features through the
open-source version later anyway?") and some people calling us twice a day,
claiming they would need instant support for Kamailio IMS (for free of
course), we sadly revoked that decision.

Currently, the Diameter-Stack in CDP is only TCP and NOT SCTP.  Of course,
you can do SIP signalling using SCTP, however this does not apply to
Diameter.

Thanks,
Carsten

--
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Millerntorplatz 1 I 20359 Hamburg I Germany
www.ng-voice.com

Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49
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Am Di., 4. Juni 2019 um 10:48 Uhr schrieb "björn bauer" :

> Hi all.
>
> The question is in addition to the this:
> https://lists.kamailio.org/pipermail/sr-users/2018-January/07.html
> At that time, Diameter over SCTP wasn't implemented in Kamailio.
>
> Carsten Bock stated in
>
> https://www.kamailio.org/events/2018-KamailioWorld/Day2/17-Carsten.Bock-Whats-Up-Kamailio-IMS-VoLTE-Kamailio.pdf
> at KamailioWorld 2018 that "SCTP support added to the Diameter-Stack".
>
> Now the question:
> How can I change the transport layer protocol of the Rx interface from
> TCP to SCTP?
>
> Regards.
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Re: [SR-Users] RTPEngine / Transcoding question

[Carsten Bock]

Hi,

the problem is, the device (a VoLTE phone) will ALWAYS offer AMR-WB and
AMR-NB and we will need to do transcoding to G722 / G711 in most cases
(unless the B-Party actually offers AMR-WB/AMR-NB).
The Answer may be pure G711 (RTPEngine will do Transcoding G711 <=> AMR-WB
[meaningless]) or G722 (RTPEngine will do Transcoding G722 <=> AMR-WB
[Makes sense]) or AMR-NB/-WB (no Transcoding [ideal]). If I could limit the
Answer from RTPEngine to contain only AMR-NB, that would ideal, as AMR-NB
has less complexity, than transcoding into AMR-WB and it would additionally
save bandwidth (which is important in our case).

However, I should mention that I am already more than happy with the
current result. Transcoding is working great, even with AMR/AMR-WB!

Thanks,
Carsten
--
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ng-voice GmbH

Millerntorplatz 1 I 20359 Hamburg I Germany
www.ng-voice.com

Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49
(0)40-52 47 593-99

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Am Mi., 22. Mai 2019 um 01:17 Uhr schrieb Richard Fuchs :

> On 21/05/2019 18.42, Carsten Bock wrote:
> > Hi Richard,
> >
> > Thanks for your reply, understood.
> >
> > The issue is following (not really an issue, more an optimization):
> > At the moment of the offer, I simply don't know, what will be
> > supported by the callee. G711 is however always supported, that's why
> > I don't want to strip it. I send the call to our upstream provider; if
> > he sends it to Telefonica, we get an answer with G722 and G711. If he
> > sends it to Deutsche Telekom, the answer is only G711. However, we
> > don't know, if it's gonna be Telefonica or Deutsche Telekom...
> >
> > I was thinking about avoiding uneccessary RTP traffic (in HD) between
> > RTPEngine and the device in case of an outbound call. From a
> > signalling perspective, I get the right result, if I use
> > sdp_remove_codecs() in my reply. However, that won't work, since the
> > RTP stream sent from RTPEngine to the device is in HD, regardless of
> > the signalling.
>
> Maybe you want to make your transcoding options also dependent on
> whether G.711 was present in the original offer, and/or in what order
> the codecs were listed? Rtpengine won't do transcoding if both sides
> support the same codec, even if other codec pairings are possible
> (unless `always-transcode` is given). Are you using that option? Are you
> using other options to reorder codecs in the offer? Because normally the
> first supported codec will be used, and rtpengine adds transcoding codec
> offers to the end of the list. So they should only be used if no other
> codec is supported by the callee, unless the callee ignores the listed
> codec preferences, or codecs were reordered, or `always-transcode` was
> used.
>
> Cheers
>
>
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[SR-Users] RTPEngine: rtpengine_manage(), Subsequent requests, Transcoding

[Carsten Bock]

Hi,

quick question:
If understood it correctly in the past (and it worked for me quite well
this way), I can simply call rtpengine_manage() for subsequent requests
without adding any additional information (e.g. used interfaces, RTP/SRTP
translation or anything like that), as this information is stored in the
internal hash table of the RTPEngine module during call setup and initial
offer/answer.

However, this does not seem to apply to the transcoding options. When I
just call "rtpengine_manage()" for a Re-INVITE of a transcoded call (in my
case AMR-WB to G722), the transcoding options are no longer applied. The
forwarded offer does not get the transcoded codecs added (e.g. in my case
G722 is not added to the list of AMR/AMR-WB from the phone).

Did I misunderstand something or is it a bug?

My RTPEngine is still on 7.2, but I will double check with more recent
versions today or tomorrow. However, I also found no commit messages
relating to this.

Thanks,
Carsten



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www.ng-voice.com

Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49
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[SR-Users] RTPEngine / Transcoding question

[Carsten Bock]

Hi,

I want to implement selective transcoding, e.g. avoiding Transcoding
between a HD codec (G722) and a non HD Codec (G711).

In case the caller does offer HD codecs, this is fine, e.g.

if(sdp_with_codecs_by_name("G722,OPUS")) {
  rtpengine_offer("codec-transcode=OPUS codec-transcode=G722
codec-transcode=PCMA");
} else {
  rtpengine_manage("codec-transcode=PCMA");
}

Now I have the issue, if the callee only sends me G711, I don't want to
offer G722 or Opus to the caller. However, rtpengine_answer() does not seem
to accept "codec-strip=G722 codec-strip=OPUS", e.g.:

onreply_route() {
  if(!sdp_with_codecs_by_name("G722,OPUS")) {
rtpengine_ manage("codec-strip=OPUS codec-strip=G722");
  }
}

As a result, the SDP always contains the transcoding options, e.g. G722 and
OPUS. I always end up in transcoding G722 to G711, which is meaningless in
that case.

Any ideas on how to solve that?
(above examples very simplified)

Thanks,
Carsten
--
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ng-voice GmbH

Millerntorplatz 1 I 20359 Hamburg I Germany
www.ng-voice.com

Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49
(0)40-52 47 593-99

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[SR-Users] TCP: Strange Device Behaviour

[Carsten Bock]

Hi,

I have a device here, which does some sort of strange things:

I have SIP Message, e.g. with a length of 1100 Bytes and the devices adds
"0" to the end to meet the MTU of the network (for a single TCP frame),
some sort of TCP padding. The following TCP Frame contains only the next
message. Looking at the trace in Wireshark, sngrep or similar, all of them
try to re-assemble the next message in the TCP stream, but however don't
decode them as SIP due to to first couple of zero's in the message.

Looking at Kamailio, it seems not to have any issues with this and
decodes/parses the message properly (they show up on Homer properly);
however it also forwards the message INCLUDING the first couple of zero's
even on UDP.

To make it complitely irritating, the phone itself would ignore such
packets, if we would forward them to the phone with starting zeros.

Has anyone seen anything like this and is it standards-compliant? Any
proper and easy way to get rid of the starting "zero's" from a message
before forwarding these messages? I could always "hack" something into
src/core/tcp_read.c, but I wonder if there is already a proper way?

Any ideas anyone?

Thanks,
Carsten

--
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ng-voice GmbH

Millerntorplatz 1 I 20359 Hamburg I Germany
www.ng-voice.com

Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49
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Re: [SR-Users] uac_replace_from and uac_auth fails to authenticate.

[Carsten Bock]

God morgen Kjeld,

You can use uac_replace_from/_to in a branch route, e.g.:

[...]
$avp(from_uri) = "sip:"+$dbr(ra=>[0,1])+"@EXTERNALIP";
t_on_branch("UPDATE_FROM");
[...]

branch_route(UPDATE_FROM) {
  uac_replace_from("$avp(from_uri)");
}

This should solve your issue.

Med venlig hilsen,
Carsten Bock (lidt Vendelbo)
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Trostbrücke 1 I 20457 Hamburg I Germany
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Managing Directors: Dr. David Bachmann, Carsten Bock


Am So., 19. Jan. 2020 um 23:20 Uhr schrieb Kjeld Flarup <
kjeld.fla...@liberalismen.dk>:

> I have a setup where I have a fallback to a GSM number
>
> I look up the GSM number and provider information in a database and sets
> the headers.
>
>dlg_manage();
>$du = "sip:" + $dbr(ra=>[0,0]);
>$tu = "sip:"+$rU+"@"+$dbr(ra=>[0,0]);
>$ru = "sip:"+$rU+"@"+$dbr(ra=>[0,0]);
> uac_replace_from("sip:"+$dbr(ra=>[0,1])+"@EXTERNALIP");
>
> After this the call goes to a failure_route to do uac_auth()
>
> Now my problem is that this works with the providers Asterisk server.
> But if the call is send to the providers Kamailio server, authentication
> is rejected.
>
> Removing uac_replace_from makes the call accepted on the Kamailio server
>
> The only possible problem I can see is that the first INVITE without
> authentication, has correct From header.
> But the second with the nonce and auth, uses the wrong From header. Thus
> two different From headers in the same SIP dialog.
>
> Unfortunately uac_replace_from is not allowed in failure_route, so I
> could test if this is the problem.
>
> Is the two different From headers a problem, and how could that be fixed.
>
>
> --
>  Med Liberalistiske Hilsner --
> Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
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>
>
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[SR-Users] PUA and db_redis

[Carsten Bock]

Hi,

has anyone been using db_redis with the Presence/PUA modules?

If yes, can you share your experience and the proper key definition?

Thanks,
Carsten
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Re: [SR-Users] s.replace Transformation and Newline

[Carsten Bock]

Hi Daniel,

Unfortunately, I already had replaced that. While I had the two \\ in
config, I also had two \\ in the SDP...

Thanks,
Carsten


Daniel-Constantin Mierla  schrieb am Mo., 18. Mai 2020,
17:49:

> Hello,
>
> I see you have two backslashes \\ for r and n. Have you tried with only
> one?
>
> \\ is supposed to be \ character.
>
> Cheers,
> Daniel
> On 18.05.20 16:32, Carsten Bock wrote:
>
> Hi,
>
> I am trying the following:
> - extract the message body
> - replace some content
> - store it in a hashtable
> - send it out in a reply in later request
>
> I do the following:
>
> [...]
> $sht(sdp=>$ci) = $(rb{s.replace,a=sendrecv,a=sendrecv\\r\\na=curr:qos
> local none});
> [...]
> set_reply_body("$sht(sdp=>$ci)", "application/sdp");
> send_reply("200", "OK");
> [...]
>
> It works like a charm, however I'm facing one small issue:
>
> My SDP looks like this:
> [...]
> a=rtpmap:8 PCMA/8000
> a=sendrecv\r\na=curr:qos local none
> a=rtcp:34949
> [...]
>
> It seems that s.replace is escaping characters.
>
> Anyone any ideas?
>
> Thanks,
> Carsten
>
>
>
>
>
>
>
>
>
> $var(x) = "abababa";
> $(var(x){s.replace,a,c} => "cbcbcbc"
> --
>
> Carsten Bock I CTO & Founder
>
> ng-voice GmbH
>
> Trostbrücke 1 I 20457 Hamburg I Germany
> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>
> Registry Office at Local Court Hamburg, HRB 120189
> Managing Directors: Dr. David Bachmann, Carsten Bock
>
> ___
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> 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
> Funding: https://www.paypal.me/dcmierla
>
>
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Re: [SR-Users] s.replace Transformation and Newline

[Carsten Bock]

Sorry for the confusion - my initial example contained a typo. The right
version:

[...]
$sht(sdp=>$ci) = $(rb{s.replace,a=sendrecv,a=sendrecv\r\na=curr:qos local
none});
[...]
set_reply_body("$sht(sdp=>$ci)", "application/sdp");
send_reply("200", "OK");
[...]

It works like a charm, however I'm facing one small issue:

My SDP looks like this:
[...]
a=rtpmap:8 PCMA/8000
a=sendrecv\r\na=curr:qos local none
a=rtcp:34949
[...]

Thanks,
Carsten

P.S.: Never write emails between two meetings...

Daniel-Constantin Mierla  schrieb am Mo., 18. Mai 2020,
18:35:

> Hello,
>
> is not clear to me: so when you had it with one backslash, it was the two
> chars '\n'  in sdp, when it was \\n was it with 2 backslashes (3 chars)?
>
> Cheers,
> Daniel
> On 18.05.20 17:58, Carsten Bock wrote:
>
> Hi Daniel,
>
> Unfortunately, I already had replaced that. While I had the two \\ in
> config, I also had two \\ in the SDP...
>
> Thanks,
> Carsten
>
>
> Daniel-Constantin Mierla  schrieb am Mo., 18. Mai
> 2020, 17:49:
>
>> Hello,
>>
>> I see you have two backslashes \\ for r and n. Have you tried with only
>> one?
>>
>> \\ is supposed to be \ character.
>>
>> Cheers,
>> Daniel
>> On 18.05.20 16:32, Carsten Bock wrote:
>>
>> Hi,
>>
>> I am trying the following:
>> - extract the message body
>> - replace some content
>> - store it in a hashtable
>> - send it out in a reply in later request
>>
>> I do the following:
>>
>> [...]
>> $sht(sdp=>$ci) = $(rb{s.replace,a=sendrecv,a=sendrecv\\r\\na=curr:qos
>> local none});
>> [...]
>> set_reply_body("$sht(sdp=>$ci)", "application/sdp");
>> send_reply("200", "OK");
>> [...]
>>
>> It works like a charm, however I'm facing one small issue:
>>
>> My SDP looks like this:
>> [...]
>> a=rtpmap:8 PCMA/8000
>> a=sendrecv\r\na=curr:qos local none
>> a=rtcp:34949
>> [...]
>>
>> It seems that s.replace is escaping characters.
>>
>> Anyone any ideas?
>>
>> Thanks,
>> Carsten
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> $var(x) = "abababa";
>> $(var(x){s.replace,a,c} => "cbcbcbc"
>> --
>>
>> Carsten Bock I CTO & Founder
>>
>> ng-voice GmbH
>>
>> Trostbrücke 1 I 20457 Hamburg I Germany
>> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>>
>> Registry Office at Local Court Hamburg, HRB 120189
>> Managing Directors: Dr. David Bachmann, Carsten Bock
>>
>> ___
>> 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
>> Funding: https://www.paypal.me/dcmierla
>>
>> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
> Funding: https://www.paypal.me/dcmierla
>
>
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[SR-Users] s.replace Transformation and Newline

[Carsten Bock]

Hi,

I am trying the following:
- extract the message body
- replace some content
- store it in a hashtable
- send it out in a reply in later request

I do the following:

[...]
$sht(sdp=>$ci) = $(rb{s.replace,a=sendrecv,a=sendrecv\\r\\na=curr:qos local
none});
[...]
set_reply_body("$sht(sdp=>$ci)", "application/sdp");
send_reply("200", "OK");
[...]

It works like a charm, however I'm facing one small issue:

My SDP looks like this:
[...]
a=rtpmap:8 PCMA/8000
a=sendrecv\r\na=curr:qos local none
a=rtcp:34949
[...]

It seems that s.replace is escaping characters.

Anyone any ideas?

Thanks,
Carsten









$var(x) = "abababa";
$(var(x){s.replace,a,c} => "cbcbcbc"
--

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ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
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Re: [SR-Users] s.replace Transformation and Newline

[Carsten Bock]

Hi Daniel,

thanks! This works!

Thanks again, Kamailio is great!
Carsten
--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Mo., 18. Mai 2020 um 22:17 Uhr schrieb Daniel-Constantin Mierla <
mico...@gmail.com>:

> Can you try with a variable:
>
> $var(x) = "a=sendrecv\r\na=curr:qos local none";
>
> and use it inside the s.replace transformation.
>
> Cheers,
> Daniel
> On 18.05.20 19:21, Carsten Bock wrote:
>
> Sorry for the confusion - my initial example contained a typo. The right
> version:
>
> [...]
> $sht(sdp=>$ci) = $(rb{s.replace,a=sendrecv,a=sendrecv\r\na=curr:qos local
> none});
> [...]
> set_reply_body("$sht(sdp=>$ci)", "application/sdp");
> send_reply("200", "OK");
> [...]
>
> It works like a charm, however I'm facing one small issue:
>
> My SDP looks like this:
> [...]
> a=rtpmap:8 PCMA/8000
> a=sendrecv\r\na=curr:qos local none
> a=rtcp:34949
> [...]
>
> Thanks,
> Carsten
>
> P.S.: Never write emails between two meetings...
>
> Daniel-Constantin Mierla  schrieb am Mo., 18. Mai
> 2020, 18:35:
>
>> Hello,
>>
>> is not clear to me: so when you had it with one backslash, it was the two
>> chars '\n'  in sdp, when it was \\n was it with 2 backslashes (3 chars)?
>>
>> Cheers,
>> Daniel
>> On 18.05.20 17:58, Carsten Bock wrote:
>>
>> Hi Daniel,
>>
>> Unfortunately, I already had replaced that. While I had the two \\ in
>> config, I also had two \\ in the SDP...
>>
>> Thanks,
>> Carsten
>>
>>
>> Daniel-Constantin Mierla  schrieb am Mo., 18. Mai
>> 2020, 17:49:
>>
>>> Hello,
>>>
>>> I see you have two backslashes \\ for r and n. Have you tried with only
>>> one?
>>>
>>> \\ is supposed to be \ character.
>>>
>>> Cheers,
>>> Daniel
>>> On 18.05.20 16:32, Carsten Bock wrote:
>>>
>>> Hi,
>>>
>>> I am trying the following:
>>> - extract the message body
>>> - replace some content
>>> - store it in a hashtable
>>> - send it out in a reply in later request
>>>
>>> I do the following:
>>>
>>> [...]
>>> $sht(sdp=>$ci) = $(rb{s.replace,a=sendrecv,a=sendrecv\\r\\na=curr:qos
>>> local none});
>>> [...]
>>> set_reply_body("$sht(sdp=>$ci)", "application/sdp");
>>> send_reply("200", "OK");
>>> [...]
>>>
>>> It works like a charm, however I'm facing one small issue:
>>>
>>> My SDP looks like this:
>>> [...]
>>> a=rtpmap:8 PCMA/8000
>>> a=sendrecv\r\na=curr:qos local none
>>> a=rtcp:34949
>>> [...]
>>>
>>> It seems that s.replace is escaping characters.
>>>
>>> Anyone any ideas?
>>>
>>> Thanks,
>>> Carsten
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> $var(x) = "abababa";
>>> $(var(x){s.replace,a,c} => "cbcbcbc"
>>> --
>>>
>>> Carsten Bock I CTO & Founder
>>>
>>> ng-voice GmbH
>>>
>>> Trostbrücke 1 I 20457 Hamburg I Germany
>>> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>>>
>>> Registry Office at Local Court Hamburg, HRB 120189
>>> Managing Directors: Dr. David Bachmann, Carsten Bock
>>>
>>> ___
>>> 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
>>> Funding: https://www.paypal.me/dcmierla
>>>
>>> --
>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
>> www.linkedin.com/in/miconda
>> Funding: https://www.paypal.me/dcmierla
>>
>> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- 
> www.linkedin.com/in/miconda
> Funding: https://www.paypal.me/dcmierla
>
>
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[SR-Users] db_redis and PUA

[Carsten Bock]

Hi,

quick question:
Did anyone ever attempt to use PUA with the new db_redis? If yes, could
someone share the definition for this?

Thanks,
Carsten
--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock
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Re: [SR-Users] BYE and TCP

[Carsten Bock]

Hi,

I think you are searching for this parameter:

http://www.kamailio.org/wiki/cookbooks/devel/core#tcp_reuse_port

It basically tells the kernel not to choose a new port at random for a new
connection, but it will try to reuse the existing port (in your case 5071).

Thanks,
Carsten


--
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ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Mi., 4. Nov. 2020 um 12:07 Uhr schrieb Kjeld Flarup <
kjeld.fla...@liberalismen.dk>:

> Hello
>
> I have a Kamailio running behind NAT, which sends calls to a VOIP
> service provider.
>
> I have setup the Kamalio to listen on port 5071, and also setup a port
> forward in the router.
>
> Now the problem is that with TCP, 5071 is not used for the dialog, but a
> new port is chosen everytime. This means that when the mobile phone
> called hands up, I never sees the BYE, because BYE is a new dialog.
>
>
> To which port is the server supposed to send the BYE, and what field
> tells the server this.
>
>
> --
>  Med Liberalistiske Hilsner --
> Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog
> Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49
> Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk
>
>
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Re: [SR-Users] R-R to MS Teams Proxy

[Carsten Bock]

Hi,

I don't understand the trouble:
Wouldn't the following lead to the exact example:

listen=tls:1.2.3.4:8007 advertise teams.tutpro.com:8007
listen=tcp:1.2.3.4:5070

As soon as you do:
$fs = "tcp:1.2.3.4:5070";
record_route();
(or put the other socket into $fs)

This results into two record-route headers, one with the hostname and one
with the IP.

We use the Hostnames in all kinds of Record-Route and Via-Headers by doing
exactly this.

Thanks,
Carsten

--
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ng-voice GmbH

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T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Do., 11. Feb. 2021 um 09:45 Uhr schrieb Juha Heinanen :

> Sergey Safarov writes:
>
> > Maybe need to move into more global changes...
> >
> > How about DNS hostname usage for all headers?
> > "Via", "Record-Route" maybe "Contact" when Kamailio with extra modules
> > play media?
> >
> > how about global directive like
> > header_with_hostname=tue
>
> I'm not sure if that is a good idea, since, for example, I would like to
> keep on using IP address 127.0.0.1:
>
> Record-Route: 
> Record-Route: 
>
> -- Juha
>
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Re: [SR-Users] [Kamailio-Business] Info: Kamailio World 2021 starting soon - live stream

[Carsten Bock]

Hi Daniel,

will the recordings of the sessions be made available at some point?
Youtube shows them to me as "private Video's" and I, unfortunately, missed
the event...

Thanks,
Carsten
--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Do., 2. Sept. 2021 um 12:51 Uhr schrieb Daniel-Constantin Mierla <
mico...@gmail.com>:

> Hello,
>
> sync'ing again the clocks around the world: the 2nd day of Kamailio
> World 2021 online edition is starting
> in about 2 hours from now (12:50 UTC Welcome and 13:00UTC the first
> session) - the agenda is available at:
>
>   * https://kamailioworld.com/k09-online/
>
> Cheers,
> Daniel
>
> On 01.09.21 12:51, Daniel-Constantin Mierla wrote:
> > Hello,
> >
> > to sync better on time, Kamailio World 2021 online edition is starting
> > in about 2 hours from now (12:50 UTC Welcome and 13:00UTC the first
> > session) - the agenda is available at:
> >
> >   * https://kamailioworld.com/k09-online/
> >
> > In short: participation is free, no registration is needed. The sessions
> > are live streamed to YouTube KamailioWorld channel:
> >
> >   * https://www.youtube.com/c/KamailioWorld/
> >
> > YouTube chat next to the live stream can be used for addressing
> > questions to speakers -- we will monitor it and ask the speaker inside
> > the video conferencing room.
> >
> > For community interaction via text chatting, you can join the Matrix
> > room #kamailio:matrix.kamailio.dev :
> >
> >   * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev
> >
> > Questions for speakers can also be addressed in this chat room.
> >
> > If anyone wants to participate in the live video conferencing room, the
> > URL is:
> >
> >   * https://meet.kamailioworld.com/live
> >
> > Just note that we may have to limit the number of participants in the
> > live session in order to ensure proper streaming quality.
> >
> > See you soon online!
> >
> > Cheers,
> > Daniel
> >
> > --
> > Daniel-Constantin Mierla -- www.asipto.com
> > www.twitter.com/miconda -- www.linkedin.com/in/miconda
> >
> --
> Daniel-Constantin Mierla -- www.asipto.com
> www.twitter.com/miconda -- www.linkedin.com/in/miconda
>
>
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Re: [SR-Users] [Kamailio-Business] Info: Kamailio World 2021 starting soon - live stream

[Carsten Bock]

Hi Daniel,

thanks! It was really a pity that I could not join

Thank you,
Carsten Bock
--
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ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Do., 9. Sept. 2021 um 13:35 Uhr schrieb Daniel-Constantin Mierla <
mico...@gmail.com>:

> The recordings of sessions with slides during Day 1 should be now youtube
> channel:
>
>   * https://www.youtube.com/c/KamailioWorld/
>
> Cheers,
> Daniel
> On 08.09.21 18:32, Julien Chavanton wrote:
>
> Thanks for sharing Daniel !
>
> On Tue, Sep 7, 2021 at 5:35 AM Giovanni Maruzzelli 
> wrote:
>
>> Thanks a lot Daniel, much appreciated (#metoo was in holidays).
>>
>> -giovanni
>>
>>
>>
>> On Tue, Sep 7, 2021 at 9:04 AM Daniel-Constantin Mierla <
>> mico...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> the recordings were made private to be cut per session -- hopefully I
>>> will get the time for finishing them during the next few days.
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 06.09.21 13:30, Daian Conrad wrote:
>>>
>>> for me show private videos too on youtube, and links during the event i
>>> can't access too.
>>>
>>> Em seg, 6 de set de 2021 05:46, Carsten Bock 
>>> escreveu:
>>>
>>>> Hi Daniel,
>>>>
>>>> will the recordings of the sessions be made available at some point?
>>>> Youtube shows them to me as "private Video's" and I, unfortunately, missed
>>>> the event...
>>>>
>>>> Thanks,
>>>> Carsten
>>>> --
>>>> Carsten Bock I CTO & Founder
>>>>
>>>> ng-voice GmbH
>>>>
>>>> Trostbrücke 1 I 20457 Hamburg I Germany
>>>> T +49 179 2021244 I www.ng-voice.com
>>>>
>>>> Registry Office at Local Court Hamburg, HRB 120189
>>>> Managing Directors: Dr. David Bachmann, Carsten Bock
>>>>
>>>>
>>>> Am Do., 2. Sept. 2021 um 12:51 Uhr schrieb Daniel-Constantin Mierla <
>>>> mico...@gmail.com>:
>>>>
>>>>> Hello,
>>>>>
>>>>> sync'ing again the clocks around the world: the 2nd day of Kamailio
>>>>> World 2021 online edition is starting
>>>>> in about 2 hours from now (12:50 UTC Welcome and 13:00UTC the first
>>>>> session) - the agenda is available at:
>>>>>
>>>>>   * https://kamailioworld.com/k09-online/
>>>>>
>>>>> Cheers,
>>>>> Daniel
>>>>>
>>>>> On 01.09.21 12:51, Daniel-Constantin Mierla wrote:
>>>>> > Hello,
>>>>> >
>>>>> > to sync better on time, Kamailio World 2021 online edition is
>>>>> starting
>>>>> > in about 2 hours from now (12:50 UTC Welcome and 13:00UTC the first
>>>>> > session) - the agenda is available at:
>>>>> >
>>>>> >   * https://kamailioworld.com/k09-online/
>>>>> >
>>>>> > In short: participation is free, no registration is needed. The
>>>>> sessions
>>>>> > are live streamed to YouTube KamailioWorld channel:
>>>>> >
>>>>> >   * https://www.youtube.com/c/KamailioWorld/
>>>>> >
>>>>> > YouTube chat next to the live stream can be used for addressing
>>>>> > questions to speakers -- we will monitor it and ask the speaker
>>>>> inside
>>>>> > the video conferencing room.
>>>>> >
>>>>> > For community interaction via text chatting, you can join the Matrix
>>>>> > room #kamailio:matrix.kamailio.dev :
>>>>> >
>>>>> >   * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev
>>>>> >
>>>>> > Questions for speakers can also be addressed in this chat room.
>>>>> >
>>>>> > If anyone wants to participate in the live video conferencing room,
>>>>> the
>>>>> > URL is:
>>>>> >
>>>>> >   * https://meet.kamailioworld.com/live
>>>>> >
>>>>> > Just note that we may have to limit the number of participants in the
>>>>> > live session in or

Re: [SR-Users] which is better for B2BUA

[Carsten Bock]

Hi,

We don't do RTP on this setup, as we only use it for connecting different
Consumer-Voice-Platforms with various upstream carriers.

We did trigger RTPEngine once from Kamailio for connecting IPv6 to IPv4
carriers, but we've removed that meanwhile.

Thanks,
Carsten

Carsten Bock I CTO & Founder




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T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock

Mojtaba  schrieb am Mo., 2. Aug. 2021, 13:04:

> Hi Carsten,
> Yes, Using SEMS as B2BUA would be good plan, But just a question, Do you
> skip RTP flow to SEMS, too?
> Thanks
>
> On Mon, Aug 2, 2021 at 1:46 PM Carsten Bock  wrote:
>
>> Hi,
>>
>> we even take it one step further:
>> Incoming <-> Kamailio <-> SEMS (B2BUA) <-> Kamailio <-> Interconnect
>>
>> We do this, to limit the required functionality of SEMS to be a B2BUA and
>> we do all further stuff (Number Portability Lookup, Least-Cost-Routing,
>> Message Manipulation, ..) on Kamailio.
>>
>> I almost hate to say this as Kamailian, but one more alternative to SEMS
>> worth checking is OpenSIPS and its b2b modules.
>>
>> Thanks,
>> Carsten
>> --
>> Carsten Bock I CTO & Founder
>>
>> ng-voice GmbH
>>
>> Trostbrücke 1 I 20457 Hamburg I Germany
>> T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com
>>
>> Registry Office at Local Court Hamburg, HRB 120189
>> Managing Directors: Dr. David Bachmann, Carsten Bock
>>
>>
>> Am Sa., 31. Juli 2021 um 23:24 Uhr schrieb Mojtaba :
>>
>>> Thanks Guys for your comments.
>>> @Gerry, I agree with you, I have a few experiences working with Yate, It
>>> could work great in enterprise solutions. Thanks
>>> @Denys, Thanks for sharing your experience, Although its configurations
>>> look complicated, But in signaling, It would work great.
>>> Just a question, Let me know which scenario is like that you use?
>>>
>>> ---
>>> or
>>> ---
>>>
>>>
>>>
>>> On Fri, Jul 30, 2021 at 9:15 AM Denys Pozniak 
>>> wrote:
>>>
>>>> Hello!
>>>> We used sems to interconnect with external operators. But with a heavy
>>>> call load and with the processing of the media stream, it often crashed.
>>>> Now we use Kamailio / Rtpengine for media stream processing and sems
>>>> for signaling processing. We install all this on one server. It probably
>>>> looks complicated, but it works great.
>>>>
>>>> вт, 27 июл. 2021 г. в 16:10, Gerry | Rigatta :
>>>>
>>>>> Hi,
>>>>>
>>>>> Some comments on Yate.
>>>>>
>>>>> Yate works well as a B2BUA. It is very performant and stable. It can
>>>>> bypass media dynamically.
>>>>>
>>>>> The main beauty of Yate is how it processes calls internally. Yate
>>>>> represents calls internally as messages with parameters (caller, callee,
>>>>> codecs …) . While the call passes from the incoming to the outgoing call
>>>>> leg the message parameters can be manipulated by different yate modules. 
>>>>> In
>>>>> turn, message parameters can trigger yate modules for some actions, e.g.
>>>>> start a sip call.
>>>>>
>>>>> The Yate internal messaging allows to build with little code complex
>>>>> call logic. E.g a mysql query result can be used directly for routing. 
>>>>> E,g,
>>>>> you can build a switch just with a couple of mysql procedures.
>>>>>
>>>>> Cheers
>>>>>
>>>>> Gerry
>>>>>
>>>>>
>>>>>
>>>>> On 27 Jul 2021, at 11:35, Karsten Horsmann 
>>>>> wrote:
>>>>>
>>>>> Hello,
>>>>>
>>>>> just for the records:
>>>>>
>>>>> You can also bypass media with FreeSWITCH if you want. The overall
>>>>> handling of many parallel calls is still not so high like in SEMS.
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media
>>>>> This is also possible on a per Call base.
>>>>>
>>>>> And you can use different profiles as well in Fr

Re: [SR-Users] which is better for B2BUA

[Carsten Bock]

Hi,

we even take it one step further:
Incoming <-> Kamailio <-> SEMS (B2BUA) <-> Kamailio <-> Interconnect

We do this, to limit the required functionality of SEMS to be a B2BUA and
we do all further stuff (Number Portability Lookup, Least-Cost-Routing,
Message Manipulation, ..) on Kamailio.

I almost hate to say this as Kamailian, but one more alternative to SEMS
worth checking is OpenSIPS and its b2b modules.

Thanks,
Carsten
--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Sa., 31. Juli 2021 um 23:24 Uhr schrieb Mojtaba :

> Thanks Guys for your comments.
> @Gerry, I agree with you, I have a few experiences working with Yate, It
> could work great in enterprise solutions. Thanks
> @Denys, Thanks for sharing your experience, Although its configurations
> look complicated, But in signaling, It would work great.
> Just a question, Let me know which scenario is like that you use?
>
> ---
> or
> ---
>
>
>
> On Fri, Jul 30, 2021 at 9:15 AM Denys Pozniak 
> wrote:
>
>> Hello!
>> We used sems to interconnect with external operators. But with a heavy
>> call load and with the processing of the media stream, it often crashed.
>> Now we use Kamailio / Rtpengine for media stream processing and sems for
>> signaling processing. We install all this on one server. It probably looks
>> complicated, but it works great.
>>
>> вт, 27 июл. 2021 г. в 16:10, Gerry | Rigatta :
>>
>>> Hi,
>>>
>>> Some comments on Yate.
>>>
>>> Yate works well as a B2BUA. It is very performant and stable. It can
>>> bypass media dynamically.
>>>
>>> The main beauty of Yate is how it processes calls internally. Yate
>>> represents calls internally as messages with parameters (caller, callee,
>>> codecs …) . While the call passes from the incoming to the outgoing call
>>> leg the message parameters can be manipulated by different yate modules. In
>>> turn, message parameters can trigger yate modules for some actions, e.g.
>>> start a sip call.
>>>
>>> The Yate internal messaging allows to build with little code complex
>>> call logic. E.g a mysql query result can be used directly for routing. E,g,
>>> you can build a switch just with a couple of mysql procedures.
>>>
>>> Cheers
>>>
>>> Gerry
>>>
>>>
>>>
>>> On 27 Jul 2021, at 11:35, Karsten Horsmann  wrote:
>>>
>>> Hello,
>>>
>>> just for the records:
>>>
>>> You can also bypass media with FreeSWITCH if you want. The overall
>>> handling of many parallel calls is still not so high like in SEMS.
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media
>>> This is also possible on a per Call base.
>>>
>>> And you can use different profiles as well in FreeSWITCH.
>>>
>>> Asterisk Experts will have more knowledge what is possible there, and
>>> maybe someone told us whats going on with Yate.
>>>
>>> Kind regards
>>> Karsten
>>>
>>> Am Di., 13. Juli 2021 um 10:11 Uhr schrieb Mojtaba :
>>>
>>>>
>>>> Hello there,
>>>> According to ,
>>>> https://lists.kamailio.org/pipermail/sr-users/2016-March/092058.html,
>>>> which talked about  B2BUA (just signalling) in Kamailio.
>>>> As i have experienced working with SEMS, freeswitch and Kamailio while
>>>> using B2BUA feature, Each of them have pros and cons:
>>>> 1- The sems is a light sip engine server with several applications
>>>> (like as sbc) for using b2bua. All incoming and outgoing calls could go to
>>>> sems server for doing b2bua like this:
>>>>
>>>>
>>>> Incoming<===>Kamailio<>Sems<>Kamailio<===>outgoing
>>>>
>>>> 2- In sems, you could disable rtp realying. It forces sems to work just
>>>> as b2bua without anchoring RTP
>>>> 3- Easy to use different active profiles in routing.
>>>>
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>>>

Re: [SR-Users] TCP Persistent Connections

[Carsten Bock]

Hi,

This is what you might be looking for:

http://www.kamailio.org/wiki/cookbooks/devel/core#tcp_reuse_port

Thanks,
Carsten

Carsten Bock I CTO & Founder




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Ross McKillop  schrieb am Mi., 10. Nov. 2021, 17:59:

> >> Any ideas? I get the feeling I'm missing something obvious!
> >
> > Perhaps this could help:
> >
> >
> https://www.kamailio.org/docs/modules/5.5.x/modules/nathelper.html#nathelper.f.add_contact_alias
>
> This involves adding additional data to the Contact URI which I want to
> avoid if possible.
>
> Also, unless I misunderstand, it seems that the alias parameter is used to
> populate the $du which I can do effectively, it's the source that I'm
> having issues with or having it use an existing socket to that same
> destination.
>
> Ross
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[SR-Users] DMQ Usrloc/Dialog - Experiences

[Carsten Bock]

Hi,

I wanted to quickly ask the group about experiences with DMQ and
Usrloc/Dialog.

I don't expect any issues when choosing such an approach, especially since
we would have re-transmits for a message (fr_timer = 30 seconds) if we did
not receive any answers or scenarios alike. So I assume, having short
"outages" (less than 30 seconds) for the communication is just fine.

So far I used DMQ in a non-IMS setup for synchronizing usrloc with ~15k
Subscribers and it worked really well.

One concern brought up (not by me) was, that synchronizing usrloc with DMQ
might not be reliable at scale. I would also love to understand how to
treat situations when the communication is lost for a longer period of time
(e.g. if the communication between nodes is lost for 5 minutes?).

Can anyone share some real-life experiences?

Thanks,
Carsten




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ng-voice GmbH

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Re: [SR-Users] DMQ Usrloc/Dialog - Experiences

[Carsten Bock]

Hi Julien,

thanks for your reply - I really appreciate your feedback and your
presentations at KamailioWorld, I hope to see you again in person soon.

Can you share some details on how many records you synchronize using DMQ
and if you ever experienced any loss of records while synchronization using
DMQ? The issue is, that on IMS the registration is typically set to 60
seconds (1 week), as we have other mechanisms to get notified if a user is
dropping out of the network (we get notifications from the LTE network
itself) and a loss of registration data would result in a user not being
reachable for a long period of time.

Thanks,
Carsten
--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Mo., 29. Nov. 2021 um 18:09 Uhr schrieb Julien Chavanton <
jchavan...@gmail.com>:

> Hi Carsten, from my experience Usrloc + DMQ works very well, rare
> replication race conditions are insignificant since the state is quite
> volatile anyway.
>
> However you can only have that many nodes, great for clustering.
>
> On Mon, Nov 29, 2021 at 6:47 AM Carsten Bock  wrote:
>
>> Hi,
>>
>> I wanted to quickly ask the group about experiences with DMQ and
>> Usrloc/Dialog.
>>
>> I don't expect any issues when choosing such an approach, especially
>> since we would have re-transmits for a message (fr_timer = 30 seconds) if
>> we did not receive any answers or scenarios alike. So I assume, having
>> short "outages" (less than 30 seconds) for the communication is just fine.
>>
>> So far I used DMQ in a non-IMS setup for synchronizing usrloc with ~15k
>> Subscribers and it worked really well.
>>
>> One concern brought up (not by me) was, that synchronizing usrloc with
>> DMQ might not be reliable at scale. I would also love to understand how to
>> treat situations when the communication is lost for a longer period of time
>> (e.g. if the communication between nodes is lost for 5 minutes?).
>>
>> Can anyone share some real-life experiences?
>>
>> Thanks,
>> Carsten
>>
>>
>>
>>
>> --
>> Carsten Bock I CTO & Founder
>>
>> ng-voice GmbH
>>
>> Trostbrücke 1 I 20457 Hamburg I Germany
>> T +49 179 2021244 I www.ng-voice.com
>>
>> Registry Office at Local Court Hamburg, HRB 120189
>> Managing Directors: Dr. David Bachmann, Carsten Bock
>> __
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Re: [SR-Users] Alternate to FoHSS for Kamailio IMS?

[Carsten Bock]

Hi,

did you check the Cx Support in Open5GS?
https://open5gs.org/open5gs/release/2021/04/09/release-v2.2.6.html

Thanks,
Carsten
--
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Am Do., 13. Jan. 2022 um 21:26 Uhr schrieb Simon,Benjamin T :

> Hello,
>
> We are attempting to setup an instance of Kamailio to serve as an IMS
> for a test setup of 4G/5G NSA with Open5GS.
>
> The instructions on the Kamailio wiki and provided by Open5GS
> currently specify to use FoHSS as the HSS solution for Kamailio IMS. As
> this HSS solution has not seen significant development in a few years
> we are reticent to deploy it on our implementation.
>
> Is there an alternative open source HSS which can be deployed to serve
> as the HSS for Kamailio instead of FoHSS, and would there need to be
> any adjustments to Kamailio to reflect the deployment of this HSS
> instead of FoHSS?
>
> Thank you,
>
>
> Benjamin Simon
>
>
> Research Assistant
> Florida Institute for Cybersecurity Research, University of Florida
> Email: bsim...@ufl.edu
>
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Re: [SR-Users] Split audio and video into different calls

[Carsten Bock]

Hi Rüdiger,

what you could achieve with Kamailio (and in my understanding, this is what
you are thinking about), is to send Audio-Calls to Group 1 (e.g. your
legacy agents) and Video-Calls to Group 2 (e.g. Agents with a Video-Phone
or Soft-Client). Is this, what you want to achieve?

Thanks,
Carsten
--
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ng-voice GmbH

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Am Fr., 13. Mai 2022 um 18:56 Uhr schrieb Rüdiger, Christoph <
christoph.ruedi...@tkelevator.com>:

> Hi everyone,
>
>
>
> we are running a call center with really good to use hardphones and audio
> only up to this point. Because of changing regulation, we soon have to deal
> with video calls as well, but only for a fraction of the calls (below 1%).
>
>
>
> Instead of switching everyone to clumsy softphones for the rare case of a
> video call, I’m exploring the ability to branch an incoming call into a
> pure audio call to the hardphone and optionally a pure video call to the
> softphone.
>
>
>
> Has anyone done this in the past and can share some experiences/ideas on
>
>1. Can it be done purely with Kamailio and rtpengine? I don’t have a
>clear picture of how to do it purely with Kamailio on the SIP side, because
>we end up with two independent legs (to the hardphone and the softphone).
>Or do I need to deploy an Asterisk, Yate, freeswitch, etc. and doing it as
>a 3-way conference where one party gets both RTP streams (the trunk) and
>the other two parties only get one RTP stream (either audio or video)?
>2. Was it worth the effort or a steady candidate for troubles?
>
>
>
> I’m looking forward to your thoughts, especially on question b).
>
>
>
> Have a nice weekend,
>
> Christoph
>
>
>
> ---
>
> Christoph Rüdiger
>
> Digital Operations Architect
> DTX OU DACH
>
> *M* +49 152 3828 0376 | christoph.ruedi...@tkelevator.com
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[SR-Users] Forward reply in onreply_route

[Carsten Bock]

Hi,

Question:
Is there a way to forward a reply from the reply route and do other stuff
afterward? We are currently doing operations on a 183 reply and it would be
useful if we could forward the reply before continuing to process other
stuff.

High-Level example:











*route {  t_on_reply("REPLY");  t_relay();}onreply_route[REPLY] {  //
Forward the reply to the Caller  t_relay(); // Does not work, as t_relay is
only allowed in request and failure-routes// Do other fancy stuff, e.g.
a DB Update or anything else}*

Any ideas?

Thanks,
Carsten




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Re: [SR-Users] Create new listener on the fly

[Carsten Bock]

Hi Hossein,

yes, you are right, it is in-efficient. It may be fine for a private
network/PoC/Lab/ or anything else small, but it will not really work for a
commercial network - for that you need something different.

Thanks,
Carsten
--
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Am Di., 25. Okt. 2022 um 23:03 Uhr schrieb H Yavari :

> Thanks Alex for your elaboration; you are right.
> This is the reason that the current ims_ipsec module will create a bunch
> of listeners at startup that looks not efficient.
>
>
> Regards,
> Hossein
>
>
> On Tuesday, October 25, 2022 at 01:28:38 PM PDT, Alex Balashov <
> abalas...@evaristesys.com> wrote:
>
>
> Well, no, it just stems from some knowledge of Kamailio’s multiprocess
> architecture.
>
> Kamailio uses a static pool of preforked SIP worker processes, and this is
> initialised once upon startup. These child processes are spawned for every
> listener, and communicate and share data using SysV IPC and shared memory.
> The shared memory pool is likewise initialised upon startup, as is the
> small fixed-size area of private memory associated with each other worker
> process (“package memory”).
>
> While perhaps not strictly impossible to alter, this setup isn’t
> particularly amenable to the dynamic creation and destruction of listeners,
> or the additional child processes they beget. The child processes need to
> be forked on startup, before their heap is laden with runtime baggage. All
> this is relatively expensive to initialise.
>
> — Alex
>
> > On Oct 25, 2022, at 4:14 PM, H Yavari  wrote:
> >
> > Thanks Alex.
> > Is there any document or material that tells more about those reasons?
> >
> >
> > Regards,
> > Hossein
> >
> >
> > On Tuesday, October 25, 2022 at 01:00:18 PM PDT, Alex Balashov <
> abalas...@evaristesys.com> wrote:
> >
> >
> > Hi,
> >
> > Unfortunately, for a variety of architectural reasons, this is not
> practical.
> >
> > — Alex
> >
> > > On Oct 25, 2022, at 12:43 PM, H Yavari  wrote:
> > >
> > > Hi Kamailio community,
> > >
> > >
> > > I am running an IPsec server beside Kamailio, so I am interested to
> know if there is any method to create a SIP listener on a specific port on
> the fly. I mean, when I create the IPsec SAs through the IPsec server, I
> ask Kamailio to create a listener/handler for that port too.
> > > This case might be very rare, but it is always good to hear community
> ideas.
> > >
> > > Thank you.
> > >
> > > Regards,
> > > Hossein
> >
> > > __
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> >
> > --
> > Alex Balashov | Principal | Evariste Systems LLC
> >
> > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> >
> >
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>
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>
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&g

Re: [SR-Users] Create new listener on the fly

[Carsten Bock]

Hi Hossein,

due to those limitations, we are using a custom, in-house solution for our
MNO/MVNO deployments, which we haven't released as open-source (yet).

Thanks,
Carsten

--
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ng-voice GmbH

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T +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Mi., 26. Okt. 2022 um 18:29 Uhr schrieb H Yavari :

> Hi Casrten,
>
> Thanks for the comment. I know you provide IMS, have you contributed to
> Kamailio IPsec, or do you have something in-house solution for this part?
> (I hope this question will not be off-topic)
>
> BR
> Hossein
>
>
> On Wednesday, October 26, 2022 at 06:37:33 AM PDT, Carsten Bock <
> cars...@ng-voice.com> wrote:
>
>
> Hi Hossein,
>
> yes, you are right, it is in-efficient. It may be fine for a private
> network/PoC/Lab/ or anything else small, but it will not really work for a
> commercial network - for that you need something different.
>
> Thanks,
> Carsten
> --
> Carsten Bock I CTO & Founder
>
> ng-voice GmbH
>
> Trostbrücke 1 I 20457 Hamburg I Germany
> T +49 179 2021244 I www.ng-voice.com
>
> Registry Office at Local Court Hamburg, HRB 120189
> Managing Directors: Dr. David Bachmann, Carsten Bock
>
>
> Am Di., 25. Okt. 2022 um 23:03 Uhr schrieb H Yavari <
> hyav...@rocketmail.com>:
>
> Thanks Alex for your elaboration; you are right.
> This is the reason that the current ims_ipsec module will create a bunch
> of listeners at startup that looks not efficient.
>
>
> Regards,
> Hossein
>
>
> On Tuesday, October 25, 2022 at 01:28:38 PM PDT, Alex Balashov <
> abalas...@evaristesys.com> wrote:
>
>
> Well, no, it just stems from some knowledge of Kamailio’s multiprocess
> architecture.
>
> Kamailio uses a static pool of preforked SIP worker processes, and this is
> initialised once upon startup. These child processes are spawned for every
> listener, and communicate and share data using SysV IPC and shared memory.
> The shared memory pool is likewise initialised upon startup, as is the
> small fixed-size area of private memory associated with each other worker
> process (“package memory”).
>
> While perhaps not strictly impossible to alter, this setup isn’t
> particularly amenable to the dynamic creation and destruction of listeners,
> or the additional child processes they beget. The child processes need to
> be forked on startup, before their heap is laden with runtime baggage. All
> this is relatively expensive to initialise.
>
> — Alex
>
> > On Oct 25, 2022, at 4:14 PM, H Yavari  wrote:
> >
> > Thanks Alex.
> > Is there any document or material that tells more about those reasons?
> >
> >
> > Regards,
> > Hossein
> >
> >
> > On Tuesday, October 25, 2022 at 01:00:18 PM PDT, Alex Balashov <
> abalas...@evaristesys.com> wrote:
> >
> >
> > Hi,
> >
> > Unfortunately, for a variety of architectural reasons, this is not
> practical.
> >
> > — Alex
> >
> > > On Oct 25, 2022, at 12:43 PM, H Yavari  wrote:
> > >
> > > Hi Kamailio community,
> > >
> > >
> > > I am running an IPsec server beside Kamailio, so I am interested to
> know if there is any method to create a SIP listener on a specific port on
> the fly. I mean, when I create the IPsec SAs through the IPsec server, I
> ask Kamailio to create a listener/handler for that port too.
> > > This case might be very rare, but it is always good to hear community
> ideas.
> > >
> > > Thank you.
> > >
> > > Regards,
> > > Hossein
> >
> > > __
> > > Kamailio - Users Mailing List - Non Commercial Discussions
> > > sr-users@lists.kamailio.org
> > > Important: keep the mailing list in the recipients, do not reply only
> to the sender!
> > > Edit mailing list options or unsubscribe:
> > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
> >
> > --
> > Alex Balashov | Principal | Evariste Systems LLC
> >
> > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> >
> >
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> > Important: keep the mailing list in the recipients, do not reply only to
> the sender!
> > Edit mailing list options o

[SR-Users] Re: IMS integration with AS

[Carsten Bock]

Hi Oleg,

This is supported by Kamailio IMS and we use it rather frequently.

However, one point is your description is not properly right:
" constructs Register message (which keeps all details of subscriber
register message) "

If you need to have all the details of the subscriber register message
(e.g. REGISTER sent by the UE), you will not find them by default in the
3rd Party registration generated by Kamailio, as it is creating a new
REGISTER (e.g. it has a new Call-ID, Contact pointing to the IMS rather
than to the UE, ...). However, you can instruct the HSS to include the
original REGISTER message as a message body or you can do a "SUBSCRIBE"
(check PUA_reginfo and related modules) for the user status in order to get
more details of the subscriber.

Thanks,
Carsten
--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Mi., 21. Dez. 2022 um 14:16 Uhr schrieb Oleg Belousov <
obelou...@gmail.com>:

> Hi.
> Can you please advise if existing ims implementation supports integration
> with ims AS (Application Server), in particular for the third party
> registration:
> - upon UE registration is completed, S-CSCF constructs Register message
> (which keeps all details of subscriber register message) and submits it to
> AS server as regular SIP-message (TS 24.229 5.4.1.7 Notification of
> Application Servers about registration status). S-CSCF obtains AS address
> from SAA (Server Assignment Answer) diameter message, received from HSS
> upon successful registration.
> --
> obelousov.tel
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Re: [SR-Users] ims ipsec connectivity

[Carsten Bock]

Hi,

be aware, that IPSec for VoLTE does not work for NAT.

Thanks,
Carsten
--
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ng-voice GmbH

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T +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock


Am Mo., 14. Nov. 2022 um 16:09 Uhr schrieb Oleg Belousov <
obelou...@gmail.com>:

> Hi, Hossein.
> Increased privileges for proxy container, now can see records in both SPD
> and SAD, but still ipsec is not established. Possible because UE is behind
> NAY - will check that either. Thanks for your advice.
> # ip xfrm state count
>  SAD count 28
> # ip xfrm policy count
>  SPD IN  16 OUT 16 FWD 0
>
> Hi, Giovanni.
> Thank you. Yes, I saw that fork and am going to try it as well. Not quite
> clear why those patches not included onto the main release.
> --
> obelousov.tel
>
>
> On Mon, Nov 14, 2022 at 11:07 AM Giovanni Maruzzelli 
> wrote:
>
>> Hello,
>>
>> you may also want to check Supreeth ipsec patches :
>>
>> https://github.com/herlesupreeth/kamailio
>>
>> -giovanni
>>
>> On Fri, Nov 11, 2022 at 1:44 PM Oleg Belousov 
>> wrote:
>>
>>> Hi,  Hossein.
>>> No, there are not. The output of these two commands is just empty.
>>> Should enable it?
>>> --
>>> obelousov.tel
>>>
>>>
>>> On Thu, Nov 10, 2022 at 9:08 PM H Yavari  wrote:
>>>
>>>> Hi Oleg,
>>>>
>>>> Can you check the ipsec SA in the OS and see that SA and policies are
>>>> there or not:
>>>> >> ip x s l
>>>> >> ip x p l
>>>>
>>>> BR,
>>>> Hossein
>>>>
>>>> On Thursday, November 10, 2022 at 03:12:34 AM PST, Oleg Belousov <
>>>> obelou...@gmail.com> wrote:
>>>>
>>>>
>>>> Hi.
>>>> Working on ims integration with the actual handset, have got a
>>>> problem with ipsec establishment to complete registration.
>>>>
>>>> Initial steps are fine, including diameter exchange and 401 (with
>>>> security server details) toward UE. On the next step UE and kamailio should
>>>> establish ipsec connection, and UE to submit the next register with a
>>>> response. As per trace UE is trying to establish the same (can see initial
>>>> TCP SYN encapsulated onto ESP), using port-s, provided in Security-Server,
>>>> but get an ICMP packet from server with destination/protocol unreachable.
>>>> No more info either in P-SCSF log, no in kern.log. Proxy is listening to
>>>> that port, it is tcp and available over telnet, so should not be a
>>>> connectivity issue.
>>>> Please let know if any ideas how to troubleshoot that further,
>>>> --
>>>> obelousov.tel
>>>> __
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>>>> sr-users@lists.kamailio.org
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>>>> to the sender!
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>>>>
>>> __
>>> Kamailio - Users Mailing List - Non Commercial Discussions
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>>> the sender!
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>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
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[SR-Users] Re: How to log expiring dialogue?

[Carsten Bock]

Hi,

You could log something in an event route:
https://kamailio.org/docs/modules/devel/modules/dialog.html#idm1691

event_route[dialog:end] {
  xlog("$dlg(h_id):$dlg(h_entry) / $dlg(callid): Call ended with - state
$dlg(state)\n")
}

That could give you at least some pointers.

Thanks,
Carsten
--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock, Quirin Maderspacher


Am Fr., 21. Apr. 2023 um 10:56 Uhr schrieb Benoît Panizzon <
benoit.paniz...@imp.ch>:

> Hi alle
>
> I think I have an issue with the dialogue and session timer module with
> Kamailio 5.4
>
> I sync the Session Expire Timer with the dialogue timeout via a common
> AVP.
>
> When I check a dialogue via kamcmd dlg.list
>
> start_ts: 1682066352
> init_ts: 1682066346
> end_ts: 0
> timeout: 1682066952
> lifetime: 600
>
> Timeout is increased on the first session refreshing re-invite, as
> expected (in this case after 300 seconds). But it looks like the
> dialogue just disappears (from kamcmd dlg.list, not from the backing
> database) when the lifetime ist reached, leaving an orphan database
> entry.
>
> I would like to digg deeper into this. Is there a way to log more
> details from the dialogue module like when it triggers dialogue expiry?
>
> --
> Mit freundlichen Grüssen
>
> -Benoît Panizzon- @ HomeOffice und normal erreichbar
> --
> I m p r o W a r e   A G-Leiter Commerce Kunden
> __
>
> Zurlindenstrasse 29 Tel  +41 61 826 93 00
> CH-4133 PrattelnFax  +41 61 826 93 01
> Schweiz Web  http://www.imp.ch
> __
> __
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[SR-Users] Re: CLI & destination based LCR

[Carsten Bock]

Hi,

I am still using the PDB Daemon with Carrierroute (initially developed by
and for 1&1) for that purpose:

https://kamailio.org/docs/modules/devel/modules/pdb.html
https://github.com/kamailio/kamailio/tree/master/utils/pdbt

It is not exactly your use case, but it allows you to match numbers to a
carrier, which then can be used for routing (e.g., have a different routing
tree for numbers from one carrier, another routing tree for numbers from
another carrier, and a default routing tree for all others).

Works like a charm.

Thanks,
Carsten


--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock, Quirin Maderspacher


Am Do., 13. Apr. 2023 um 15:48 Uhr schrieb Juha Heinanen :

> John Cahill writes:
>
> > I'd like to build an LCR solution that routes on both CLI & destination
> e.g.
> >
> > CLI matches ^\+49*, destinations matches ^\+44* => route to gateway 5
> >
> > for a large rule set (~ tens of thousands). Any recommendations for
> > which modules to use?
>
> Check if LCR module scales for our use case.  It does support your
> routing requirement:
>
>   A gateway matches a request if user part of Request-URI matches
>   "prefix", caller URI matches "from_uri" pattern, caller URI userpart
>   matches mtree with "mt_value", and callee URI matches "request_uri"
>   pattern in a tuple that is associated with the gateway.
>
> -- Juha
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[SR-Users] Re: DB_CLUSTER failure with carrierroute (and others?)

[Carsten Bock]

Hi,

It could also be the case, that carrierroute uses Rawquery, which is not
supported by all database modules. However, when using db_cluster, the
db_cluster module claims to support "any" DB operation, even though the
underlying db-module may not support all DB operations.

Thanks,
Carsten

Henning Westerholt  schrieb am Mi., 8. Feb. 2023, 19:07:

> Hello,
>
>
>
> if you can not reproduce it with db_mysql, it might be indeed some issue
> specific related to sqlite. Executing the tests with debug log level should
> give you more information about the internal database operations, in case
> you like to look more into it.
>
>
>
> Cheers,
>
>
>
> Henning
>
>
>
> --
>
> Henning Westerholt – https://skalatan.de/blog/
>
> Kamailio services – https://gilawa.com
>
>
>
> *From:* Kaufman 
> *Sent:* Wednesday, February 8, 2023 6:04 PM
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* [SR-Users] Re: DB_CLUSTER failure with carrierroute (and
> others?)
>
>
>
> I cannot reproduce this issue against mysql, so perhaps it is just
> something related to abstraction and concurrent access with sqlite.
>
>
>
> *Kaufman*
>
>
>
> *From:* Kaufman 
> *Sent:* Wednesday, February 8, 2023 10:48 AM
> *To:* Kamailio (SER) - Users Mailing List 
> *Subject:* [SR-Users] DB_CLUSTER failure with carrierroute (and others?)
>
>
>
> All,
>
>
>
> I’ve been trying to chase down an issue with db_cluster and permissions in
> combination with something else, where with my full configuration using the
> same cluster in multiple modules has failures.  I’ve been trying to create
> a simplified configuration to consistently reproduce the problem without
> much success, however I did stumble upon THIS particular problem.  The
> configuration here works fine as is – avpops uses the cluster URL, and
> carrierroute uses the direct SQLite URL.  If I define WITH_DBCLUSTER, then
> carrierroute fails fetching rows using the cluster connection.  Note that
> AVPOPS is only used as a general test to see if the cluster config works.
> If I remove references to avpops , carrierroute still fails loading from
> the cluster URL.   Thoughts?
>
>
>
>
>
>
>
>
>
> #!KAMAILIO
>
>
>
> loadmodule "db_sqlite"
>
> loadmodule "pv"
>
> loadmodule "db_cluster"
>
> loadmodule "carrierroute"
>
> loadmodule "avpops"
>
>
>
> modparam("db_cluster", "connection", "db1=>sqlite:etc/db.sqlite")
>
> modparam("db_cluster", "cluster", "cl1=>db1=9s9s")
>
> modparam("db_cluster", "inactive_interval", 30)
>
>
>
> #!ifdef WITH_DBCLUSTER
>
> modparam("carrierroute", "db_url", "cluster://cl1")
>
> #!else
>
> modparam("carrierroute", "db_url", "sqlite:etc/db.sqlite")
>
> #!endif
>
> modparam("carrierroute", "config_source", "db")
>
>
>
> modparam("avpops", "db_url", "cluster://cl1")
>
> modparam("avpops", "avp_table", "usr_preferences")
>
>
>
> request_route {
>
> exit;
>
> }
>
>
>
>
>
>
> Failure log:
>
>
>
> 0(1) ERROR: carrierroute [cr_db.c:345]: load_route_data_db(): Fetching
> rows failed
>
> 0(1) ERROR: carrierroute [cr_data.c:179]: reload_route_data(): could not
> load routing data
>
> 0(1) ERROR: carrierroute [carrierroute.c:240]: mod_init(): could not
> prepare route data
>
> 0(1) ERROR:  [core/sr_module.c:942]: init_mod(): Error while
> initializing module carrierroute (/usr/lib/kamailio/modules/carrierroute.so)
>
>
>
> *Kaufman*
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[SR-Users] Re: Kamailio and MS Teams side by side present staus und chat function

[Carsten Bock]

Hi,

You should investigate the API of MS Teams. MS Teams has an API to set the
status of a subscriber - I haven't used it myself yet, but I've checked its
existence:
-
https://learn.microsoft.com/en-us/graph/api/presence-setpresence?view=graph-rest-1.0=http

I haven't checked yet on chat messages, but an API for messaging exists:
-
https://learn.microsoft.com/en-us/graph/api/chatmessage-post?view=graph-rest-1.0=http

This could be the glue to put it together.

Maybe this could server as some pointers.

Thanks,
Carsten
--
Carsten Bock I CTO & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 179 2021244 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock, Quirin Maderspacher


Am Mi., 31. Mai 2023 um 07:27 Uhr schrieb Hans-Jürgen Brand <
hans-juergen.br...@gmx.net>:

> I got a request to test Kamailio and MS Teams side by side at the same
> customer.
> IT department would like to use Kamailio with Blink SIP client and the HR
> and Management department would like to use MS Teams.
>
> Makeing VIOP calls are not the problem. But how to exchange the present
> status and exchange chat messages?
> In Kamailio we use SIP SIMPEL / XCAP.
>
> Does someone knows a solution? As fare as I know Microsoft has removed the
> Xmpp Gateway function in the solution.
>
> kind regards
> Hans-Jürgen Brand
>
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[SR-Users] Re: Software bill of materials (SBOM)

[Carsten Bock via sr-users]

Hi,

We are using that "ldd" approach for our Docker containers: We are running
ldd on the Kamailio binary and the modules from config (may vary -
depending on system) and use that result to create a slim Kamailio
Container "from scratch" - without any operating system.

Thanks,
Carsten


--
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ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 1511 5942983 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock, Quirin Maderspacher


Am Do., 28. Sept. 2023 um 19:22 Uhr schrieb Daniel-Constantin Mierla via
sr-users :

>
> On 28.09.23 13:13, Olle E. Johansson via sr-users wrote:
>
>
>
> On 28 Sep 2023, at 12:36, Ivan Ribakov via sr-users
>   wrote:
>
> Hi Olle,
>
> Yes, I realised by now that taking enabled Kamailio modules into account
> when generating SBOM is too much to ask. I'd be ok with obtaining full list
> of Kamailio dependencies (with transitive dependencies if possible) and
> then manually filtering them based on module usage. Not sure if at any
> point during Kamailio build process all sources + dependency
> sources/binaries are present in the system for scanning/identification?
>
> I'm mainly interested in listing (and validating licenses) and having a
> general inventory. Any recommendations?
>
> I did try a beta of a tool in cyclonedx toolset for scanning C files and
> it crashed. Will try again, but so far I haven’t succeeded.
> I suggest we would need one SBOM based on a linux distro, like Debian and
> one
> more generic based on C code and the versions of libraries we recommend. I
> have tried to add pointers to the various
> third party dependencies in the READMEs over the years in a somewhat
> unstructured effort, but the information is there.
> Maybe we can add the dependencies in a way that’s parseable in order to
> build an SBOM.
>
> C code doesn’t have package management like Python, Perl, Go and others so
> it’s tricky to automate creation of SBOMs.
>
> I think that the SBOM tree for the source code and dependencies would grow
> quite large.
>
> Anyway -  at this time, I failed. :-)
>
> Maybe leveraging ldd in a first phase can help building the chain of
> dependencies:
>
> $ ldd src/kamailio
> linux-vdso.so.1 (0x91745000)
> libm.so.6 => /lib/aarch64-linux-gnu/libm.so.6 (0x90f3)
> libc.so.6 => /lib/aarch64-linux-gnu/libc.so.6 (0x90d8)
> /lib/ld-linux-aarch64.so.1 (0x9170c000)
>
> $ ldd src/modules/tls/tls.so
> linux-vdso.so.1 (0x96e5d000)
> libssl.so.3 => /lib/aarch64-linux-gnu/libssl.so.3 (0x96ca)
> libcrypto.so.3 => /lib/aarch64-linux-gnu/libcrypto.so.3
> (0x968b)
> libc.so.6 => /lib/aarch64-linux-gnu/libc.so.6 (0x9670)
> /lib/ld-linux-aarch64.so.1 (0x96e24000)
>
> $ ldd /lib/aarch64-linux-gnu/libcrypto.so.3
> linux-vdso.so.1 (0x9952c000)
> libc.so.6 => /lib/aarch64-linux-gnu/libc.so.6 (0x98f5)
> /lib/ld-linux-aarch64.so.1 (0x994f3000)
>
> Might take some time, a matter of what modules are used, but if really
> needed, the process should be doable manually.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla (@ asipto.com)twitter.com/miconda -- 
> linkedin.com/in/miconda
> Kamailio Consultancy and Development Services
> Kamailio Advanced Training - Online - Nov 14-16, 2023 -- asipto.com
>
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[SR-Users] Re: dlg_req_within kamailio 5.7 CSeq values

[Carsten Bock via sr-users]

Hi,

I wrote it ;-)

You could check a branch with some changes regarding CSeq Handling:
https://github.com/kamailio/kamailio/tree/carstenbock/dialog_cseq_update/src/modules

I use my branch to send frequent in-dialog INFO messages (USSD,
https://github.com/carstenbock/ussd-poc), which works fine. I haven't found
the time yet to do more testing, so I haven't done a PR for it yet.

Thanks,
Carsten

--
Carsten Bock I Chief Technology Innovation Officer & Founder

ng-voice GmbH

Trostbrücke 1 I 20457 Hamburg I Germany
T +49 1511 5942983 I www.ng-voice.com

Registry Office at Local Court Hamburg, HRB 120189
Managing Directors: Dr. David Bachmann, Carsten Bock, Quirin Maderspacher


Am Mo., 13. Nov. 2023 um 11:46 Uhr schrieb Daniel-Constantin Mierla via
sr-users :

> Hello,
> On 10.11.23 15:57, Jonathan Hunter via sr-users wrote:
>
>
>
> Hi All,
>
>
>
> I am playing around with the dlg_req_within to send  a REINVITE to toggle
> some rtpengine parameters and this works fine.
>
>
>
> The first time I use the command after tracking the dialog, it sends a
> REINVITE with an incremented CSEQ value.
>
>
>
> However when I issue it again, it sends another REINVITE but with the same
> CSEQ value, so again 2 for example.
>
>
>
> How can you ensure its incremented or can you set it somehow?
>
>
>
> As would like to generate at least 2 reinvites per dialog.
>
> I haven't implemented this function, so I am not sure if it leverages it,
> but the dialog module has a modparam to enable cseq tracking, which is used
> by uac module for auth functionality. If you haven't set that parameter,
> try with it and see if it does the magic.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla (@ asipto.com)twitter.com/miconda -- 
> linkedin.com/in/miconda
> Kamailio Consultancy and Development Services
>
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[SR-Users] Re: Kamailio - handover between LTE and Wifi

[Carsten Bock via sr-users]

Hi,

The handover between VoWifi and VoLTE is fairly simple, as it has little to
do with IMS.
Technically speaking, the data session is migrated from the PGW (LTE) to
the ePDG (Wifi), using GTP on the S2b interface. The device keeps it's IP
address and the the handover is fully transparent to the IMS and related
systems.
If someone is at MWC in a little more than a week and wants to chat, drop
me a message.

Thanks,
Carsten

Christian Hahn via sr-users  schrieb am Do.,
15. Feb. 2024, 09:41:

> Hi all,
>
>
> While researching for Open Source IMS Services I’v found the presentation
> “What’s up with IMS & VoLTE?” at KamailioWorld 2018.
>
>
>
> There is a quote that raises a question:
>
> “Things, you shouldn’t expect (at least initially):
>
> ● Seamless handover between LTE and Wifi, as this requires some
> integration with the PGW of the LTE”
>
>
>
> My question is: Has the VoLTE <-> VoWiFi Handover been implemented in the
> meantime?
>
>
> BR Christian
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