[SR-Users] Re: Failover with dialog dmq while dialog state 2

[Henning Westerholt via sr-users]

Hello,

thanks for the detailed e-mail. As also indicated in the module documentation, 
the dialog module DMQ replication will not replicate everything, its main 
use-case is for profile data sharing. 
https://kamailio.org/docs/modules/5.8.x/modules/dialog.html#dialog.p.enable_dmq

In the past months there have been some other discussions on the users lists 
about similar scenarios (I think related to billing/accounting) and dialog with 
DMQ, which might be interesting for you in this regard.

If you find issues where the DMQ synchronisation is lacking some functionality 
in the dialog module, you can create a feature request in our issue tracker. 
There is of course no guarantee that this limitation is also timely addressed.

Regarding the INVITE and CANCEL scenario, this is usually not related to dialog 
but to the tm module. As you also mentioned, there is no replication of 
transaction state in tm.

Cheers,

Henning

-- 
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

> -Original Message-
> From: Björn Klasen via sr-users 
> Sent: Dienstag, 7. Mai 2024 11:18
> To: Kamailio (SER) - Users Mailing List 
> Cc: Björn Klasen 
> Subject: [SR-Users] Failover with dialog dmq while dialog state 2
> 
> Hello @all
> 
> as we want to enhance our VoIP-Plattform with DMQ functionalities for
> replication of Userlocation, Dialog and Hash Tables I made a lot of tests 
> durings
> the last week.
> 
> For testing I use two Debian 12 Server with Kamailio 5.7.4 (latest git)
> 
> Both server have their own floating IP so in case one server fails the 
> floating
> changes to the working server. This is done via keepalived
> 
> On both servers Kamailio is active and use ip_free_bind=1 so both Kamailio
> instances can listen to all necessary IPs.
> 
> So we have following network configuration:
> 
> Server1:
> static IP: 192.168.253.201
> floating IP: 192.168.253.200
> 
> Server2:
> static IP: 192.168.253.202
> floating IP: 192.168.253.210
> 
> Now I test some calls while I trigger a failover, so Server1 takes over
> 192.168.253.10
> 
> Message
> Test
> Result
> CustomerA -> Proxy -> CustomerB
> Failover while Dialogstate 4
> BYE is relayed correctly. Dialog is changing to state 5 CDR needs to 
> be
> written manually -> no problem at all CustomerA -> Proxy -> CustomerB
> Failover while Dialogstate 2
> 200 OK is relayed correctly although no transaction exist an 
> therefore no
> t_reply route is triggered. This is not a big deal because i can use default
> onreply_route so all necessary variables can be used via hash table.
> BUT Big Problem: Dialog state stays 2, so a following ACK is a bogus event and
> the dialog state is never changed to 4. Even worse: after 300 seconds the
> dialog is cleaned away by the timer, so I have no chance to do any kind of
> accounting.
> 
> Test 2 is a real mess, because for me there is only one solution. I have to 
> check
> after a failover on incoming replies to an INVITE that was handled on Server2 
> if
> these current dialog state is < 4 and cancel them.
> But I can't see how to do it? Are there other ways than using t_cancel?
> Or is it even some kind of a bug that a 200 OK does not trigger state 3 on
> dialog (although I have in mind that it has to be something to do with missing
> transaction data, that are not replicated because of missing replication
> capabilities of tm module)?
> If none of the above is possible I really asking myself of a real use case of
> dialog dmq replication.
> 
> --
> Björn Klasen, Teamleitung NGN VoIP-Backbone TNG Stadtnetz GmbH, TNG-
> Technik Gerhard-Fröhler-Straße 12
> 24106 Kiel・Deutschland
> 
> T +49 431 7097-10
> F +49 431 7097-555
> bkla...@tng.de
> https://www.tng.de
> 
> Executive board (Geschäftsführer):
> Dr. Sven Willert (CEO/Vorsitz),
> Gunnar Peter, Sven Schade,
> Carsten Tolkmit, Bernd Sontheimer
> 
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> können Sie unter https://www.tng.de/datenschutz/ abrufen.
> ___
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[SR-Users] Re: [EXTERNAL] Kamailio and RTPengine and no voice on calls

[Richard Fuchs via sr-users]

On 09/05/2024 10.16, palany via sr-users wrote:


I have an AWS ec2 instance with Kamailio and rtpengine configured. I 
am having a challenge of no voice on calls. I have sip client on my 
mobile phone and soft phone on my laptop and the two are on different 
networks behind nat. Signaling is working fine but there is no media 
and when I check on on the SDP the call logs are showing private ips 
instead of public ips : o=- 3 2 IN IP4 192.168.1.33 and c=IN IP4 
172.31.36.50. I understand this is a NAT issue but I am failing to see 
where is the challenge. My rtpengine seems to be working fine .


Most likely you need to use the "advertised address" notation in your 
interface config as on AWS generally you don't have the public IP 
address bound on a local interface. So something like


interface = external/172.31.36.50!13.246.88.123;internal/172.31.36.50

Cheers
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[SR-Users] Kamailio and RTPengine and no voice on calls

[palany via sr-users]

I have an AWS ec2 instance with Kamailio and rtpengine configured. I am
having a challenge of no voice on calls. I have sip client on my mobile
phone and soft phone on my laptop and the two are on different networks
behind nat. Signaling is working fine but there is no media and when I check
on on the SDP the call logs are showing private ips instead of public ips :
o=- 3 2 IN IP4 192.168.1.33 and c=IN IP4 172.31.36.50. I understand this is
a NAT issue but I am failing to see where is the challenge. My rtpengine
seems to be working fine .

 

Below is my Kamailio.cfg file link : 

 

https://docs.google.com/document/d/1MKZJGlQjxYbdtw6QcCm9_EmvVPOkMs-YXjlq1ZM2
mmY/edit?usp=drive_link

 

PCAP file for the call:

 

https://docs.google.com/document/d/12gofLYhN0-lYIZORngIeuaYxLudLjiNdZgY504Ei
XFs/edit?usp=drive_link

 

rtpengine.conf

 

https://docs.google.com/document/d/1QPxey49UzuN3Wd8jp8jc0mFhh31rr1alIO-gl4gP
JEI/edit?usp=drive_link

 

 

Please help.

 

 

 

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[SR-Users] How to set "set_dialog_profile" value for Leg-a and leg-b.

[satyaprakash ch via sr-users]

Hi,

We are using kamailio and freeswitch, So when we make a call from A to B
the call is going via "kamailio -> freeswitch" and "freeswitch ->
kamailio". Please suggest me how i can set "set_dialog_profile" for both
legs with unique value in kamailio config file.

Thanks.
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[SR-Users] Re: Is there any way to modify the Refer-To header username?

[Shah Hussain Khattak via sr-users]

Hi Henning,

Thanks a lot for the suggestion, it helped me to fix my issue. I used the 
replace function and it worked perfectly for my use-case!

4.7.  replace(re, txt)
Replaces the first occurrence of re with txt.
Meaning of the parameters is as follows:

  *
re - Regular expression.
  *
txt - String.

This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, 
BRANCH_ROUTE.
Example 1.7. replace usage

...
replace("server", "Kamailio SIP Proxy");
...



Regards,
Shah Hussain

From: Henning Westerholt 
Sent: Thursday, May 9, 2024 5:36 PM
To: Kamailio (SER) - Users Mailing List 
Cc: Shah Hussain Khattak 
Subject: RE: Is there any way to modify the Refer-To header username?


Hello,



there are other functions in textops module, e.g. subst(..) that can be used to 
do replacements.



Have a look to this module for more details and examples. The approach of 
deleting and adding it new works as well, of course.



Cheers,



Henning



--

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com





From: Shah Hussain Khattak via sr-users 
Sent: Donnerstag, 9. Mai 2024 08:16
To: Kamailio (SER) - Users Mailing List 
Cc: Shah Hussain Khattak 
Subject: [SR-Users] Is there any way to modify the Refer-To header username?



Hello Experts,



I wanted to check, is there any way to modify the username part of the Refer-To 
header in Kamailio? For example, I am getting the following header:



REFER-TO: >



the username part is missing.



Can I add a username inside it? For example, change it to



REFER-TO: >

REFER-TO: >



I could not find any solution in the mailing list, the only way was to remove 
the header and then re-add it.



Thank you!



Regards,

Shah Hussain










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[SR-Users] Re: Is there any way to modify the Refer-To header username?

[Henning Westerholt via sr-users]

Hello,

there are other functions in textops module, e.g. subst(..) that can be used to 
do replacements.

Have a look to this module for more details and examples. The approach of 
deleting and adding it new works as well, of course.

Cheers,

Henning

--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://gilawa.com


From: Shah Hussain Khattak via sr-users 
Sent: Donnerstag, 9. Mai 2024 08:16
To: Kamailio (SER) - Users Mailing List 
Cc: Shah Hussain Khattak 
Subject: [SR-Users] Is there any way to modify the Refer-To header username?

Hello Experts,

I wanted to check, is there any way to modify the username part of the Refer-To 
header in Kamailio? For example, I am getting the following header:

REFER-TO: >

the username part is missing.

Can I add a username inside it? For example, change it to

REFER-TO: >
REFER-TO: >

I could not find any solution in the mailing list, the only way was to remove 
the header and then re-add it.

Thank you!

Regards,
Shah Hussain





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[SR-Users] Is there any way to modify the Refer-To header username?

[Shah Hussain Khattak via sr-users]

Hello Experts,

I wanted to check, is there any way to modify the username part of the Refer-To 
header in Kamailio? For example, I am getting the following header:

REFER-TO: 

the username part is missing.

Can I add a username inside it? For example, change it to

REFER-TO: 
REFER-TO: 


I could not find any solution in the mailing list, the only way was to remove 
the header and then re-add it.

Thank you!

Regards,
Shah Hussain






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