Re: [OpenSIPS-Users] ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown call-id

[Prathibha B]

Trying to bind the socket for port = '36412'
rtpengine[7938]: DEBUG: [2fanirs85upoeuj766as]: [core] Something already
keeps this port, trying to take another port(s)

On Tue, 19 Sept 2023 at 14:38, Prathibha B  wrote:

> Can anyone pls help?
>
> On Tue, 19 Sept 2023 at 13:39, Prathibha B 
> wrote:
>
>> I am getting this error:
>>
>> ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown
>> call-id
>>
>> How to resolve this?
>>
>> I've attached the opensips.cf file with this email.
>>
>> --
>> Regards,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 traffic

[Kosmas Palios]

My apologies for responding to the wrong thread. Please ignore my last message, 
starting with:

"Thank you for your response.

We have tried generating ..."

Sent from Outlook for iOS

From: Bogdan-Andrei Iancu 
Sent: Tuesday, September 19, 2023 12:13:51 PM
To: OpenSIPS users mailling list ; Kosmas Palios 
; Tahir Almas Dhesi 
Subject: Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 
traffic

Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bog...@opensips.org. 
Μάθετε γιατί είναι σημαντικό
Not sure what the callcenter has to do here with the LB scenario. LB is simply 
distributing (according to the load) a call to a bunch of similar back servers.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/19/23 11:58 AM, Kosmas Palios wrote:
Thank you for your response.

We have tried generating different configs. We suspect that the problem is in 
the call center scenario (possibly the absence of). There seems to be a problem 
in connecting the incoming call to a new client instance.

The b2b mod generated INVITE that has the right agent in the INVITE Header but 
the wrong  Header. We have looked into the debug messages and there seem to 
be no reason for this issue. I will post some useful debug lines here later.

Sent from Outlook for iOS

From: Users 
 on 
behalf of Bogdan-Andrei Iancu 
Sent: Tuesday, September 19, 2023 11:46:01 AM
To: OpenSIPS users mailling list 
; Tahir Almas Dhesi 

Subject: Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 
traffic


Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
bog...@opensips.org. Μάθετε γιατί είναι 
σημαντικό

Hi,

Take a look here, it might be a good starting point:
https://www.opensips.org/Documentation/Generating-Configs-3-2

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/5/23 3:22 PM, Tahir Almas Dhesi wrote:
Interested to know how we can deploy opensips as load balancer for  T.38 calls 
, whether it will work fine or we need to use rabbitmq for load balancing

scenario is

ICTFax => opesips load balancer => freeswitch nodes



regards
Tahir Almas

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT




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Re: [OpenSIPS-Users] Wrong TCP socket being used on TLS registrations

[Bogdan-Andrei Iancu]

Hi Ray,

Do you use any TCP aliasing options in your cfg ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/2/23 3:17 AM, Ray Jackson wrote:


Hi all,

I'm facing a weird issue which I think is related to broken TCP socket 
reuse logic where the wrong client is receiving incoming calls due to 
the wrong socket being used for the incoming INVITE.


The scenario is when I have 2 clients registering using TLS behind NAT 
at the same Public IPv4 address and both clients are using the same 
private port number.  So client 1 registers and the Via and contact 
header looks like:


Via: SIP/2.0/TLS 
192.168.42.162:5062;branch=z9hG4bK1409895926;rport;alias Contact: 
;reg-id=2;+sip.instance=""


Client 2 registers from behind the same Public IPv4 address and the 
Via and contact header looks like:


Via: SIP/2.0/TLS 192.168.42.186:5062;branch=z9hG4bK-aff1f3b3 Contact: 
;expires=300


The location table shows Client 1 received field of 103.212.1.2:5062 
and Client 103.212.1.2:23456


When a call comes in for Client 1 the location lookup seems to return 
the correct 'received' address and port (e.g. 103.212.1.2:5062) and 
all the logs indicate that this is where the SIP INVITE *should* be 
going to (in the $du field).  However when you check the SIP traffic 
it selects Client 2's socket and the traffic goes to port 23456 
instead of 5062.


I think this is related somehow to the TCP port reuse logic inside 
Opensips.  My suspicion is that Opensips is looking at the Contact or 
Via port number (which is the same for both client 1 and 2) and then 
somehow mapping this to the wrong TCP received socket.


Does anybody have any suggestions here?  Should I be fixing the NAT in 
the Contact header (using fix_nated_contact).  I read somewhere that 
you shouldn't rewrite the Contact header to avoid problems with 
sending a different Contact URI to the client on calls.  Or is this 
issue more related to the Via header and the TCP port reuse logic 
looking at this port instead of the actual received port when choosing 
the outgoing socket?


FYI: I am using both force_rport() and fix_nated_register() for 
incoming registrations from these clients and matching_mode of 0 in 
usrloc.  However, I am not using fix_nated_contact() for registrations.


Thanks,

Ray



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Re: [OpenSIPS-Users] rest_post logs

[Bogdan-Andrei Iancu]

Thanks Alberto for the update here.

YEs, that was my suspicion also, that the logs are generated by the lib 
itself and not by the opensips code. We may check if the curl lib gives 
us any possibility to control its verbosity - could you open a Feature 
Request here https://github.com/OpenSIPS/opensips/issues - we use this 
but only for coding related reports, not for community help in using 
OpenSIPS.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/19/23 12:14 PM, Alberto wrote:

Hi,

Thanks for your reply,

I'm using v3.2.
I realized that that's the output from libcurl, which I can turn off 
by setting log_stdout and log_stderror to no.


Thanks again

p.s. Have you ever thought about a forum instead of the mailing list? 
I would have gone back and answered/closed my own question, but I 
didn't know if I should have done that via mail.



On Tue, 19 Sept 2023 at 09:57, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


HI Alberto,

What OpenSIPS version do you have?

And are those the only log lines you get ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
   https://www.siphub.com  

On 9/1/23 4:43 PM, Alberto wrote:

Hi,

Is there a way to turn off logs for rest_post requests?

This is my log settings:
log_level=-1
xlog_level=-1
log_stdout=yes
log_stderror=yes
log_facility=LOG_LOCAL0

This is the type of output I want to suppress:
Sep 01 13:31:48 opensips opensips[59740]: > POST
/api/opensips/doit HTTP/1.1
Sep 01 13:31:48 opensips opensips[59740]: Host: 127.0.0.1
Sep 01 13:31:48 opensips opensips[59740]: Accept: */*
Sep 01 13:31:48 opensips opensips[59740]: Content-Type:
application/json
Sep 01 13:31:48 opensips opensips[59740]: Content-Length: 156
Sep 01 13:31:48 opensips opensips[59740]:
Sep 01 13:31:48 opensips opensips[59740]: * upload completely
sent off: 156 out of 156 bytes
Sep 01 13:31:48 opensips opensips[59740]: * Mark bundle as not
supporting multiuse
Sep 01 13:31:48 opensips opensips[59740]: < HTTP/1.1 200 OK
Sep 01 13:31:48 opensips opensips[59740]: < Server: nginx
Sep 01 13:31:48 opensips opensips[59740]: < Date: Fri, 01 Sep
2023 13:31:48 GMT
Sep 01 13:31:48 opensips opensips[59740]: < Content-Type:
application/json; charset=utf-8
Sep 01 13:31:48 opensips opensips[59740]: < Content-Length: 360
Sep 01 13:31:48 opensips opensips[59740]: < Connection: keep-alive
Sep 01 13:31:48 opensips opensips[59740]: <
Access-Control-Allow-Origin: *
Sep 01 13:31:48 opensips opensips[59740]: <
Content-Security-Policy: default-src 'self';base-uri
'self';font-src 'self' https: data:;form-action
'self';frame-ancestors 'self';img-src 'self' data:;object-src
'none';script-src 'self';script-src-attr 'none';style-src 'self'
https: 'unsafe-inline';upgrade-insecure-requests
Sep 01 13:31:48 opensips opensips[59740]: <
Cross-Origin-Opener-Policy: same-origin
Sep 01 13:31:48 opensips opensips[59740]: <
Cross-Origin-Resource-Policy: same-origin
Sep 01 13:31:48 opensips opensips[59740]: < Origin-Agent-Cluster: ?1
Sep 01 13:31:48 opensips opensips[59740]: < Referrer-Policy:
no-referrer
Sep 01 13:31:48 opensips opensips[59740]: <
Strict-Transport-Security: max-age=15552000; includeSubDomains
Sep 01 13:31:48 opensips opensips[59740]: <
X-Content-Type-Options: nosniff
Sep 01 13:31:48 opensips opensips[59740]: <
X-DNS-Prefetch-Control: off
Sep 01 13:31:48 opensips opensips[59740]: < X-Download-Options:
noopen
Sep 01 13:31:48 opensips opensips[59740]: < X-Frame-Options:
SAMEORIGIN
Sep 01 13:31:48 opensips opensips[59740]: <
X-Permitted-Cross-Domain-Policies: none
Sep 01 13:31:48 opensips opensips[59740]: < X-XSS-Protection: 0
Sep 01 13:31:48 opensips opensips[59740]: <
Sep 01 13:31:48 opensips opensips[59740]: * Connection #0 to host
127.0.0.1 left intact

Thanks

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Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2

[Bogdan-Andrei Iancu]

Thanks, let's move the discussion over there.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/19/23 11:47 AM, Kosmas Palios wrote:

of course, here it is

https://github.com/OpenSIPS/opensips/issues/3176 



thank you



Sent from Outlook for iOS 

*From:* Bogdan-Andrei Iancu 
*Sent:* Tuesday, September 19, 2023 11:42:03 AM
*To:* OpenSIPS users mailling list ; Kosmas 
Palios ; Kosmas Palios 

*Subject:* Re: [OpenSIPS-Users] Call center issue in 3.1; call center 
not working after migration to 3.2



Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
bog...@opensips.org. Μάθετε γιατί είναι σημαντικό 




For the second issue, could you point the GH ticket?

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
   https://www.siphub.com  
On 9/19/23 11:35 AM, Kosmas Palios wrote:
This is indeed the solution, as we figured out by ourselves and as 
documented in the tm documentation!


Thank you for your kind reply here, as well as at the issue I posted 
sometime ago i github (my username is kosniaz).


Any input to the second issue (raised on github as well about 10 days 
ago) would be greatly appreciated!


Thank you very much one more time for your insights.

Have a great day!

Kosmas

Sent from Outlook for iOS 

*From:* Users  
 on behalf of Bogdan-Andrei 
Iancu  

*Sent:* Tuesday, September 19, 2023 11:20:23 AM
*To:* OpenSIPS users mailling list  
; Kosmas Palios 
 
*Subject:* Re: [OpenSIPS-Users] Call center issue in 3.1; call center 
not working after migration to 3.2



Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
bog...@opensips.org . Μάθετε γιατί είναι 
σημαντικό 



Hi Kosmas,

For Issue 1, try placing a t_newtran(); before calling the cc 
function in the script - this will prevent the issues due to 
retransmissions.


Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
   https://www.siphub.com  
On 7/4/23 1:15 PM, Kosmas Palios wrote:


Hello community,


Our team has managed to setup a SIP Trunk to forward calls to a 
number of SIP clients, in opensips 3.1.16. We are using the call 
center module, and it works fine for low traffic. We would like to 
ask two separate but related questions.



ISSUE 1


We are using the call center module to forward calls to 100 SIP 
agents, and it works well if traffic is relatively low (about 25 
incoming calls per minute). However, when traffic is higher, i.e. up 
to 60 incoming calls per minute, we see calls getting rejected 
because of cc_handle_call() failing with error message:



DBG:b2b_entities:server_new: It is a retransmission, drop

ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b 
server instance



Unfortunately, every time this happens, an agent's status gets stuck 
to "incall" forever, even though no cc_calls row includes him. So 
that agent is lost.



We are running in UDP mode, using 6 UDP workers. I’m attaching the 
configuration file as opensips_3_1_16.cfg


I can share the whole setup if needed.


ISSUE 2


We decided to migrate to 3.2 after seeing the bugfix to b2b_clients 
leak. When we got to migrating the call center, we read this 
blogpost: 
https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/ 




" When comes to the modules using the b2b_logicAPI (providing 
features on top of the B2B engine), the only affected one is the 
call_centermodule. The change is minor – the xml file controlling 
the call queuing logic was removed, as not needed any more. 
Otherwise, in terms of usage, it is exactly the same."



However, when we removed the lines:


modparam("b2b_logic_xml","script_scenario", 
"/etc/opensips/scenario_callcenter.xml")


modparam("call_center", "b2b_scenario", "call center")


the call center started behaving weird: it created another invite to 
the sip trunk, instead of creating the invite to the agent (the call 
id was good, but the to uri was wrong). I can give detailed logs on 
this, but I wouldn't want to make this email any bigger than it 
already is. I’m also attaching the configuration file as 
opensips_3_2_13.cfg



To sum up, our questions are:


1. Any ideas on what the problem is 

Re: [OpenSIPS-Users] rest_post logs

[Alberto]

Hi,

Thanks for your reply,

I'm using v3.2.
I realized that that's the output from libcurl, which I can turn off by
setting log_stdout and log_stderror to no.

Thanks again

p.s. Have you ever thought about a forum instead of the mailing list? I
would have gone back and answered/closed my own question, but I didn't know
if I should have done that via mail.


On Tue, 19 Sept 2023 at 09:57, Bogdan-Andrei Iancu 
wrote:

> HI Alberto,
>
> What OpenSIPS version do you have?
>
> And are those the only log lines you get ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 9/1/23 4:43 PM, Alberto wrote:
>
> Hi,
>
> Is there a way to turn off logs for rest_post requests?
>
> This is my log settings:
> log_level=-1
> xlog_level=-1
> log_stdout=yes
> log_stderror=yes
> log_facility=LOG_LOCAL0
>
> This is the type of output I want to suppress:
> Sep 01 13:31:48 opensips opensips[59740]: > POST /api/opensips/doit
> HTTP/1.1
> Sep 01 13:31:48 opensips opensips[59740]: Host: 127.0.0.1
> Sep 01 13:31:48 opensips opensips[59740]: Accept: */*
> Sep 01 13:31:48 opensips opensips[59740]: Content-Type: application/json
> Sep 01 13:31:48 opensips opensips[59740]: Content-Length: 156
> Sep 01 13:31:48 opensips opensips[59740]:
> Sep 01 13:31:48 opensips opensips[59740]: * upload completely sent off:
> 156 out of 156 bytes
> Sep 01 13:31:48 opensips opensips[59740]: * Mark bundle as not supporting
> multiuse
> Sep 01 13:31:48 opensips opensips[59740]: < HTTP/1.1 200 OK
> Sep 01 13:31:48 opensips opensips[59740]: < Server: nginx
> Sep 01 13:31:48 opensips opensips[59740]: < Date: Fri, 01 Sep 2023
> 13:31:48 GMT
> Sep 01 13:31:48 opensips opensips[59740]: < Content-Type:
> application/json; charset=utf-8
> Sep 01 13:31:48 opensips opensips[59740]: < Content-Length: 360
> Sep 01 13:31:48 opensips opensips[59740]: < Connection: keep-alive
> Sep 01 13:31:48 opensips opensips[59740]: < Access-Control-Allow-Origin: *
> Sep 01 13:31:48 opensips opensips[59740]: < Content-Security-Policy:
> default-src 'self';base-uri 'self';font-src 'self' https:
> data:;form-action 'self';frame-ancestors 'self';img-src 'self'
> data:;object-src 'none';script-src 'self';script-src-attr
> 'none';style-src 'self' https: 'unsafe-inline';upgrade-insecure-requests
> Sep 01 13:31:48 opensips opensips[59740]: < Cross-Origin-Opener-Policy:
> same-origin
> Sep 01 13:31:48 opensips opensips[59740]: < Cross-Origin-Resource-Policy:
> same-origin
> Sep 01 13:31:48 opensips opensips[59740]: < Origin-Agent-Cluster: ?1
> Sep 01 13:31:48 opensips opensips[59740]: < Referrer-Policy: no-referrer
> Sep 01 13:31:48 opensips opensips[59740]: < Strict-Transport-Security:
> max-age=15552000; includeSubDomains
> Sep 01 13:31:48 opensips opensips[59740]: < X-Content-Type-Options: nosniff
> Sep 01 13:31:48 opensips opensips[59740]: < X-DNS-Prefetch-Control: off
> Sep 01 13:31:48 opensips opensips[59740]: < X-Download-Options: noopen
> Sep 01 13:31:48 opensips opensips[59740]: < X-Frame-Options: SAMEORIGIN
> Sep 01 13:31:48 opensips opensips[59740]: <
> X-Permitted-Cross-Domain-Policies: none
> Sep 01 13:31:48 opensips opensips[59740]: < X-XSS-Protection: 0
> Sep 01 13:31:48 opensips opensips[59740]: <
> Sep 01 13:31:48 opensips opensips[59740]: * Connection #0 to host
> 127.0.0.1 left intact
>
> Thanks
>
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 traffic

[Bogdan-Andrei Iancu]

Not sure what the callcenter has to do here with the LB scenario. LB is 
simply distributing (according to the load) a call to a bunch of similar 
back servers.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/19/23 11:58 AM, Kosmas Palios wrote:

Thank you for your response.

We have tried generating different configs. We suspect that the 
problem is in the call center scenario (possibly the absence of). 
There seems to be a problem in connecting the incoming call to a new 
client instance.


The b2b mod generated INVITE that has the right agent in the INVITE 
Header but the wrong  Header. We have looked into the debug 
messages and there seem to be no reason for this issue. I will post 
some useful debug lines here later.


Sent from Outlook for iOS 

*From:* Users  on behalf of 
Bogdan-Andrei Iancu 

*Sent:* Tuesday, September 19, 2023 11:46:01 AM
*To:* OpenSIPS users mailling list ; Tahir 
Almas Dhesi 
*Subject:* Re: [OpenSIPS-Users] opensips deployment as load balancer 
for T.38 traffic



Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
bog...@opensips.org. Μάθετε γιατί είναι σημαντικό 




Hi,

Take a look here, it might be a good starting point:
https://www.opensips.org/Documentation/Generating-Configs-3-2 



Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
   https://www.siphub.com  
On 9/5/23 3:22 PM, Tahir Almas Dhesi wrote:
Interested to know how we can deploy opensips as load balancer for 
T.38 calls , whether it will work fine or we need to use rabbitmq for 
load balancing


scenario is

ICTFax => opesips load balancer => freeswitch nodes



regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com 
Leveraging open source in ICT


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Re: [OpenSIPS-Users] upgrade failed

[Bogdan-Andrei Iancu]

Hi,

That;s not the recommended way to fix the issue, as the the structure of 
the `dispatcher ` table is actually different in 3.4 and 3.1. Bypassing 
the check may result in failing DB queries.


The proper fix is install the correct dispatcher table for 3.4:
https://github.com/OpenSIPS/opensips/blob/3.4/scripts/mysql/dispatcher-create.sql

Or use the db migration/create functionality of opensips-cli

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/10/23 10:02 AM, Social Boh wrote:


mysql -u root -p

use opensips

update version set table_version='9' where table_name='dispatcher';

quit

---
I'm SoCIaL, MayBe
El 10/09/2023 a las 12:33 a. m., Saint Michael escribió:

I attempted an upgrade to version 3.4 from 3.1 and It failed
ERROR:core:db_check_table_version: invalid version 8 for table 
dispatcher found, expected 9

ERROR:dispatcher:mod_init: failed to init database support
ERROR:core:init_mod: failed to initialize module dispatcher
ERROR:core:main: error while initializing modules

how do I upgrade the dispatcher table from 8 version 8 to version 9?
The script is fine.




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Re: [OpenSIPS-Users] ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown call-id

[Prathibha B]

Can anyone pls help?

On Tue, 19 Sept 2023 at 13:39, Prathibha B  wrote:

> I am getting this error:
>
> ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown
> call-id
>
> How to resolve this?
>
> I've attached the opensips.cf file with this email.
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 traffic

[Kosmas Palios]

Thank you for your response.

We have tried generating different configs. We suspect that the problem is in 
the call center scenario (possibly the absence of). There seems to be a problem 
in connecting the incoming call to a new client instance.

The b2b mod generated INVITE that has the right agent in the INVITE Header but 
the wrong  Header. We have looked into the debug messages and there seem to 
be no reason for this issue. I will post some useful debug lines here later.

Sent from Outlook for iOS

From: Users  on behalf of Bogdan-Andrei Iancu 

Sent: Tuesday, September 19, 2023 11:46:01 AM
To: OpenSIPS users mailling list ; Tahir Almas Dhesi 

Subject: Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 
traffic

Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bog...@opensips.org. 
Μάθετε γιατί είναι σημαντικό
Hi,

Take a look here, it might be a good starting point:
https://www.opensips.org/Documentation/Generating-Configs-3-2

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/5/23 3:22 PM, Tahir Almas Dhesi wrote:
Interested to know how we can deploy opensips as load balancer for  T.38 calls 
, whether it will work fine or we need to use rabbitmq for load balancing

scenario is

ICTFax => opesips load balancer => freeswitch nodes



regards
Tahir Almas

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT




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Re: [OpenSIPS-Users] rest_post logs

[Bogdan-Andrei Iancu]

HI Alberto,

What OpenSIPS version do you have?

And are those the only log lines you get ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/1/23 4:43 PM, Alberto wrote:

Hi,

Is there a way to turn off logs for rest_post requests?

This is my log settings:
log_level=-1
xlog_level=-1
log_stdout=yes
log_stderror=yes
log_facility=LOG_LOCAL0

This is the type of output I want to suppress:
Sep 01 13:31:48 opensips opensips[59740]: > POST /api/opensips/doit 
HTTP/1.1

Sep 01 13:31:48 opensips opensips[59740]: Host: 127.0.0.1
Sep 01 13:31:48 opensips opensips[59740]: Accept: */*
Sep 01 13:31:48 opensips opensips[59740]: Content-Type: application/json
Sep 01 13:31:48 opensips opensips[59740]: Content-Length: 156
Sep 01 13:31:48 opensips opensips[59740]:
Sep 01 13:31:48 opensips opensips[59740]: * upload completely sent 
off: 156 out of 156 bytes
Sep 01 13:31:48 opensips opensips[59740]: * Mark bundle as not 
supporting multiuse

Sep 01 13:31:48 opensips opensips[59740]: < HTTP/1.1 200 OK
Sep 01 13:31:48 opensips opensips[59740]: < Server: nginx
Sep 01 13:31:48 opensips opensips[59740]: < Date: Fri, 01 Sep 2023 
13:31:48 GMT
Sep 01 13:31:48 opensips opensips[59740]: < Content-Type: 
application/json; charset=utf-8

Sep 01 13:31:48 opensips opensips[59740]: < Content-Length: 360
Sep 01 13:31:48 opensips opensips[59740]: < Connection: keep-alive
Sep 01 13:31:48 opensips opensips[59740]: < Access-Control-Allow-Origin: *
Sep 01 13:31:48 opensips opensips[59740]: < Content-Security-Policy: 
default-src 'self';base-uri 'self';font-src 'self' https: 
data:;form-action 'self';frame-ancestors 'self';img-src 'self' 
data:;object-src 'none';script-src 'self';script-src-attr 
'none';style-src 'self' https: 'unsafe-inline';upgrade-insecure-requests
Sep 01 13:31:48 opensips opensips[59740]: < 
Cross-Origin-Opener-Policy: same-origin
Sep 01 13:31:48 opensips opensips[59740]: < 
Cross-Origin-Resource-Policy: same-origin

Sep 01 13:31:48 opensips opensips[59740]: < Origin-Agent-Cluster: ?1
Sep 01 13:31:48 opensips opensips[59740]: < Referrer-Policy: no-referrer
Sep 01 13:31:48 opensips opensips[59740]: < Strict-Transport-Security: 
max-age=15552000; includeSubDomains
Sep 01 13:31:48 opensips opensips[59740]: < X-Content-Type-Options: 
nosniff

Sep 01 13:31:48 opensips opensips[59740]: < X-DNS-Prefetch-Control: off
Sep 01 13:31:48 opensips opensips[59740]: < X-Download-Options: noopen
Sep 01 13:31:48 opensips opensips[59740]: < X-Frame-Options: SAMEORIGIN
Sep 01 13:31:48 opensips opensips[59740]: < 
X-Permitted-Cross-Domain-Policies: none

Sep 01 13:31:48 opensips opensips[59740]: < X-XSS-Protection: 0
Sep 01 13:31:48 opensips opensips[59740]: <
Sep 01 13:31:48 opensips opensips[59740]: * Connection #0 to host 
127.0.0.1 left intact


Thanks

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Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2

[Kosmas Palios]

of course, here it is

https://github.com/OpenSIPS/opensips/issues/3176

thank you



Sent from Outlook for iOS

From: Bogdan-Andrei Iancu 
Sent: Tuesday, September 19, 2023 11:42:03 AM
To: OpenSIPS users mailling list ; Kosmas Palios 
; Kosmas Palios 
Subject: Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working 
after migration to 3.2

Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bog...@opensips.org. 
Μάθετε γιατί είναι σημαντικό
For the second issue, could you point the GH ticket?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/19/23 11:35 AM, Kosmas Palios wrote:
This is indeed the solution, as we figured out by ourselves and as documented 
in the tm documentation!

Thank you for your kind reply here, as well as at the issue I posted sometime 
ago i github (my username is kosniaz).

Any input to the second issue (raised on github as well about 10 days ago) 
would be greatly appreciated!

Thank you very much one more time for your insights.

Have a great day!

Kosmas

Sent from Outlook for iOS

From: Users 
 on 
behalf of Bogdan-Andrei Iancu 
Sent: Tuesday, September 19, 2023 11:20:23 AM
To: OpenSIPS users mailling list 
; Kosmas Palios 

Subject: Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working 
after migration to 3.2


Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
bog...@opensips.org. Μάθετε γιατί είναι 
σημαντικό

Hi Kosmas,

For Issue 1, try placing a t_newtran(); before calling the cc function in the 
script - this will prevent the issues due to retransmissions.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 7/4/23 1:15 PM, Kosmas Palios wrote:

Hello community,


Our team has managed to setup a SIP Trunk to forward calls to a number of SIP 
clients, in opensips 3.1.16. We are using the call center module, and it works 
fine for low traffic. We would like to ask two separate but related questions.


ISSUE 1


We are using the call center module to forward calls to 100 SIP agents, and it 
works well if traffic is relatively low (about 25 incoming calls per minute). 
However, when traffic is higher, i.e. up to 60 incoming calls per minute, we 
see calls getting rejected because of cc_handle_call() failing with error 
message:


DBG:b2b_entities:server_new: It is a retransmission, drop

ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server 
instance


Unfortunately, every time this happens, an agent's status gets stuck to 
"incall" forever, even though no cc_calls row includes him. So that agent is 
lost.


We are running in UDP mode, using 6 UDP workers. I’m attaching the 
configuration file as opensips_3_1_16.cfg

I can share the whole setup if needed.


ISSUE 2


We decided to migrate to 3.2 after seeing the bugfix to b2b_clients leak. When 
we got to migrating the call center, we read this blogpost: 
https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/


" When comes to the modules using the b2b_logic API (providing features on top 
of the B2B engine), the only affected one is the call_center module. The change 
is minor – the xml file controlling the call queuing logic was removed, as not 
needed any more. Otherwise, in terms of usage, it is exactly the same."


However, when we removed the lines:


modparam("b2b_logic_xml","script_scenario", 
"/etc/opensips/scenario_callcenter.xml")

modparam("call_center", "b2b_scenario", "call center")


the call center started behaving weird: it created another invite to the sip 
trunk, instead of creating the invite to the agent (the call id was good, but 
the to uri was wrong). I can give detailed logs on this, but I wouldn't want to 
make this email any bigger than it already is. I’m also attaching the 
configuration file as opensips_3_2_13.cfg


To sum up, our questions are:


1. Any ideas on what the problem is with creating a new server instance for 
high numbers of calls?

2. What's the recommended way to migrate the call center to version 3.2 ? Can 
we find an example script-driven call center somewhere?

Thank you in advance for your help!

Best regards,

Kosmas



P.S.: about our team: we are a small team from Athens, Greece integrating voice 
assistants on various platforms. Unfortunately we missed the latest Opensips 
summit held last September.




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Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 traffic

[Bogdan-Andrei Iancu]

Hi,

Take a look here, it might be a good starting point:
    https://www.opensips.org/Documentation/Generating-Configs-3-2

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/5/23 3:22 PM, Tahir Almas Dhesi wrote:
Interested to know how we can deploy opensips as load balancer for 
T.38 calls , whether it will work fine or we need to use rabbitmq for 
load balancing


scenario is

ICTFax => opesips load balancer => freeswitch nodes



regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com 
Leveraging open source in ICT


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Re: [OpenSIPS-Users] Increased macro MAX_BRANCHES and behavior of tm module

[Bogdan-Andrei Iancu]

Hi,

The 31 is a hard limit as the TM module is internally using an integer 
for storing a branch bitmask. Here is the limitation coming from. The 
change is not a trivial one, still not impossible (to drop the int 
bitmask and move to an array of int)


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/7/23 2:49 PM, Dmitry Ponomaryov wrote:


Have a nice day, everyone

Question is to increase the value of MAX_BRANCHES[1] to 32, for 
example, which will go beyond the boundaries of the source code in 
tm.c, namely condition[2], which clearly shows that only 30 is 
possible, how critical it is to change the condition in tm.c, in order 
to get the need value? [1] 
https://github.com/OpenSIPS/opensips/blob/master/config.h#L169 
 [2] 
https://github.com/OpenSIPS/opensips/blob/master/modules/tm/tm.c#L817-L821 




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Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2

[Bogdan-Andrei Iancu]

For the second issue, could you point the GH ticket?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/19/23 11:35 AM, Kosmas Palios wrote:
This is indeed the solution, as we figured out by ourselves and as 
documented in the tm documentation!


Thank you for your kind reply here, as well as at the issue I posted 
sometime ago i github (my username is kosniaz).


Any input to the second issue (raised on github as well about 10 days 
ago) would be greatly appreciated!


Thank you very much one more time for your insights.

Have a great day!

Kosmas

Sent from Outlook for iOS 

*From:* Users  on behalf of 
Bogdan-Andrei Iancu 

*Sent:* Tuesday, September 19, 2023 11:20:23 AM
*To:* OpenSIPS users mailling list ; Kosmas 
Palios 
*Subject:* Re: [OpenSIPS-Users] Call center issue in 3.1; call center 
not working after migration to 3.2



Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
bog...@opensips.org. Μάθετε γιατί είναι σημαντικό 




Hi Kosmas,

For Issue 1, try placing a t_newtran(); before calling the cc function 
in the script - this will prevent the issues due to retransmissions.


Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
   https://www.siphub.com  
On 7/4/23 1:15 PM, Kosmas Palios wrote:


Hello community,


Our team has managed to setup a SIP Trunk to forward calls to a 
number of SIP clients, in opensips 3.1.16. We are using the call 
center module, and it works fine for low traffic. We would like to 
ask two separate but related questions.



ISSUE 1


We are using the call center module to forward calls to 100 SIP 
agents, and it works well if traffic is relatively low (about 25 
incoming calls per minute). However, when traffic is higher, i.e. up 
to 60 incoming calls per minute, we see calls getting rejected 
because of cc_handle_call() failing with error message:



DBG:b2b_entities:server_new: It is a retransmission, drop

ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b 
server instance



Unfortunately, every time this happens, an agent's status gets stuck 
to "incall" forever, even though no cc_calls row includes him. So 
that agent is lost.



We are running in UDP mode, using 6 UDP workers. I’m attaching the 
configuration file as opensips_3_1_16.cfg


I can share the whole setup if needed.


ISSUE 2


We decided to migrate to 3.2 after seeing the bugfix to b2b_clients 
leak. When we got to migrating the call center, we read this 
blogpost: 
https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/ 




" When comes to the modules using the b2b_logicAPI (providing 
features on top of the B2B engine), the only affected one is the 
call_centermodule. The change is minor – the xml file controlling the 
call queuing logic was removed, as not needed any more. Otherwise, in 
terms of usage, it is exactly the same."



However, when we removed the lines:


modparam("b2b_logic_xml","script_scenario", 
"/etc/opensips/scenario_callcenter.xml")


modparam("call_center", "b2b_scenario", "call center")


the call center started behaving weird: it created another invite to 
the sip trunk, instead of creating the invite to the agent (the call 
id was good, but the to uri was wrong). I can give detailed logs on 
this, but I wouldn't want to make this email any bigger than it 
already is. I’m also attaching the configuration file as 
opensips_3_2_13.cfg



To sum up, our questions are:


1. Any ideas on what the problem is with creating a new server 
instance for high numbers of calls?


2. What's the recommended way to migrate the call center to version 
3.2 ? Can we find an example script-driven call center somewhere?



Thank you in advance for your help!

Best regards,

Kosmas


P.S.: about our team: we are a small team from Athens, Greece 
integrating voice assistants on various platforms. Unfortunately we 
missed the latest Opensips summit held last September.




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Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2

[Kosmas Palios]

This is indeed the solution, as we figured out by ourselves and as documented 
in the tm documentation!

Thank you for your kind reply here, as well as at the issue I posted sometime 
ago i github (my username is kosniaz).

Any input to the second issue (raised on github as well about 10 days ago) 
would be greatly appreciated!

Thank you very much one more time for your insights.

Have a great day!

Kosmas

Sent from Outlook for iOS

From: Users  on behalf of Bogdan-Andrei Iancu 

Sent: Tuesday, September 19, 2023 11:20:23 AM
To: OpenSIPS users mailling list ; Kosmas Palios 

Subject: Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working 
after migration to 3.2

Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bog...@opensips.org. 
Μάθετε γιατί είναι σημαντικό
Hi Kosmas,

For Issue 1, try placing a t_newtran(); before calling the cc function in the 
script - this will prevent the issues due to retransmissions.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 7/4/23 1:15 PM, Kosmas Palios wrote:

Hello community,


Our team has managed to setup a SIP Trunk to forward calls to a number of SIP 
clients, in opensips 3.1.16. We are using the call center module, and it works 
fine for low traffic. We would like to ask two separate but related questions.


ISSUE 1


We are using the call center module to forward calls to 100 SIP agents, and it 
works well if traffic is relatively low (about 25 incoming calls per minute). 
However, when traffic is higher, i.e. up to 60 incoming calls per minute, we 
see calls getting rejected because of cc_handle_call() failing with error 
message:


DBG:b2b_entities:server_new: It is a retransmission, drop

ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server 
instance


Unfortunately, every time this happens, an agent's status gets stuck to 
"incall" forever, even though no cc_calls row includes him. So that agent is 
lost.


We are running in UDP mode, using 6 UDP workers. I’m attaching the 
configuration file as opensips_3_1_16.cfg

I can share the whole setup if needed.


ISSUE 2


We decided to migrate to 3.2 after seeing the bugfix to b2b_clients leak. When 
we got to migrating the call center, we read this blogpost: 
https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/


" When comes to the modules using the b2b_logic API (providing features on top 
of the B2B engine), the only affected one is the call_center module. The change 
is minor – the xml file controlling the call queuing logic was removed, as not 
needed any more. Otherwise, in terms of usage, it is exactly the same."


However, when we removed the lines:


modparam("b2b_logic_xml","script_scenario", 
"/etc/opensips/scenario_callcenter.xml")

modparam("call_center", "b2b_scenario", "call center")


the call center started behaving weird: it created another invite to the sip 
trunk, instead of creating the invite to the agent (the call id was good, but 
the to uri was wrong). I can give detailed logs on this, but I wouldn't want to 
make this email any bigger than it already is. I’m also attaching the 
configuration file as opensips_3_2_13.cfg


To sum up, our questions are:


1. Any ideas on what the problem is with creating a new server instance for 
high numbers of calls?

2. What's the recommended way to migrate the call center to version 3.2 ? Can 
we find an example script-driven call center somewhere?

Thank you in advance for your help!

Best regards,

Kosmas



P.S.: about our team: we are a small team from Athens, Greece integrating voice 
assistants on various platforms. Unfortunately we missed the latest Opensips 
summit held last September.




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Re: [OpenSIPS-Users] is_myself()

[Bogdan-Andrei Iancu]

The documentation is your best friend,:
https://www.opensips.org/Documentation/Script-CoreFunctions-3-2#is_myself

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 9/15/23 8:37 AM, Prathibha B wrote:

I've changed
*from_uri!=myself to !is_myself($fu)*
*!uri==myself to !is_myself($ru)*
*!uri==myself to !is_myself($ru)*

*Is it correct?
*

--
Regards,
B.Prathibha

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Re: [OpenSIPS-Users] Implementation Issues with Opensips as a Load Balancer

[Bogdan-Andrei Iancu]

Hi Joan,

A typically LB does not handle registrations, is just doing pass thru 
for the calls. Again the LB is just balancing the calls between multiple 
back servers, it does not handle REGISTER traffic


Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 7/10/23 9:22 PM, Joan Leyrana wrote:


Hello Team, I'm trying to implement Opensips with Asterisk as a load 
balancer (I have been looking at the documentation, but I can't find a 
guide for the newer versions of Opensips). I have already installed 
the MySQL database, Opensips version 3.3, and OCP 9.3.3. I have added 
my Asterisk servers to the load balancer, generated my load balancer 
script, and activated the necessary modules in the config file. 
However, it is not working for me, and I have the following doubts:


Will the VoIP phones register with Opensips? Why does generating the 
load balancer script disable the registrar module?


The route logic generated by the load balancer is not working, or at 
least it doesn't allow me to register the phones.


My understanding is that the route is as follows:

VoIP Phone -- >Registrar --> Opensips Opensips --> Media --> Asterisk

Am I skipping any important steps

Regards,


Banner 
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Gerente Infraestructura | Nextor Telecom
** *O *55 4440 6008
*E *j...@nextor.io 
Presa Falcón 128 Irrigación, CDMX 11500
*www.nextor.io* 



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Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2

[Bogdan-Andrei Iancu]

Hi Kosmas,

For Issue 1, try placing a t_newtran(); before calling the cc function 
in the script - this will prevent the issues due to retransmissions.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com

On 7/4/23 1:15 PM, Kosmas Palios wrote:


Hello community,


Our team has managed to setup a SIP Trunk to forward calls to a number 
of SIP clients, in opensips 3.1.16. We are using the call center 
module, and it works fine for low traffic. We would like to ask two 
separate but related questions.



ISSUE 1


We are using the call center module to forward calls to 100 SIP 
agents, and it works well if traffic is relatively low (about 25 
incoming calls per minute). However, when traffic is higher, i.e. up 
to 60 incoming calls per minute, we see calls getting rejected because 
of cc_handle_call() failing with error message:



DBG:b2b_entities:server_new: It is a retransmission, drop

ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b 
server instance



Unfortunately, every time this happens, an agent's status gets stuck 
to "incall" forever, even though no cc_calls row includes him. So that 
agent is lost.



We are running in UDP mode, using 6 UDP workers. I’m attaching the 
configuration file as opensips_3_1_16.cfg


I can share the whole setup if needed.


ISSUE 2


We decided to migrate to 3.2 after seeing the bugfix to b2b_clients 
leak. When we got to migrating the call center, we read this blogpost: 
https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/ 




" When comes to the modules using the b2b_logicAPI (providing features 
on top of the B2B engine), the only affected one is the 
call_centermodule. The change is minor – the xml file controlling the 
call queuing logic was removed, as not needed any more. Otherwise, in 
terms of usage, it is exactly the same."



However, when we removed the lines:


modparam("b2b_logic_xml","script_scenario", 
"/etc/opensips/scenario_callcenter.xml")


modparam("call_center", "b2b_scenario", "call center")


the call center started behaving weird: it created another invite to 
the sip trunk, instead of creating the invite to the agent (the call 
id was good, but the to uri was wrong). I can give detailed logs on 
this, but I wouldn't want to make this email any bigger than it 
already is. I’m also attaching the configuration file as 
opensips_3_2_13.cfg



To sum up, our questions are:


1. Any ideas on what the problem is with creating a new server 
instance for high numbers of calls?


2. What's the recommended way to migrate the call center to version 
3.2 ? Can we find an example script-driven call center somewhere?



Thank you in advance for your help!

Best regards,

Kosmas


P.S.: about our team: we are a small team from Athens, Greece 
integrating voice assistants on various platforms. Unfortunately we 
missed the latest Opensips summit held last September.




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[OpenSIPS-Users] ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown call-id

[Prathibha B]

I am getting this error:

ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown
call-id

How to resolve this?

I've attached the opensips.cf file with this email.

-- 
Regards,
B.Prathibha
### Global Parameters #

#debug_mode=yes

log_level=3
xlog_level=3
log_stderror=no
log_facility=LOG_LOCAL0

udp_workers=4

#disable_dns_blacklist=no

#dns_try_ipv6=yes


socket=udp:172.31.34.24:5060 as 65.2.167.22:5060  # CUSTOMIZE ME
socket=tcp:172.31.34.24:5060 as 65.2.167.22:5060  # CUSTOMIZE ME
socket=tls:172.31.34.24:5061 as 65.2.167.22:5061  # CUSTOMIZE ME
socket=ws:172.31.34.24:8080 as 65.2.167.22:8080
socket=wss:172.31.34.24:7443 as 65.2.167.22:7443

### Modules Section 

#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"

loadmodule "proto_udp.so"
loadmodule "proto_tcp.so"
loadmodule "proto_tls.so"
loadmodule "proto_ws.so"
loadmodule "proto_wss.so"
modparam("proto_wss", "wss_port", 7443)
modparam("proto_wss", "wss_max_msg_chunks", 16)

 SIGNALING module
loadmodule "signaling.so"

 StateLess module
loadmodule "sl.so"

 Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

 Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

 MAX ForWarD module
loadmodule "maxfwd.so"

 SIP MSG OPerationS module
loadmodule "sipmsgops.so"

 FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)

 MYSQL module
loadmodule "db_mysql.so"

 HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", )

 USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "working_mode_preset", "single-instance-sql-write-back")
modparam("usrloc", "db_url",
"mysql://root:root@localhost/opensips") # CUSTOMIZE ME

 avpops module
loadmodule "avpops.so"
modparam("avpops", "db_url",
"mysql://root:root@localhost/opensips") # CUSTOMIZE ME


 REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
modparam("registrar", "received_avp", "$avp(received_nh)")
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)

 ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)

modparam("acc", "detect_direction", 0)
modparam("acc", "db_url",
"mysql://root:root@localhost/opensips") # CUSTOMIZE ME

 AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
"mysql://root:root@localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")

loadmodule "alias_db.so"
modparam("alias_db", "db_url",
"mysql://root:root@localhost/opensips") # CUSTOMIZE ME

 DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
"mysql://root:root@localhost/opensips") # CUSTOMIZE ME

  NAT modules
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:pinger@127.0.0.1") #CUSTOMIZE ME
modparam("nathelper", "received_avp", "$avp(received_nh)")

  DIALPLAN module
loadmodule "dialplan.so"
modparam("dialplan", "db_url",
"mysql://root:root@localhost/opensips") # CUSTOMIZE ME

  MI_HTTP module
loadmodule "mi_http.so"

loadmodule "proto_udp.so"
loadmodule "proto_tcp.so" 

loadmodule "tls_openssl.so"
#loadmodule "tls_wolfssl.so"
loadmodule "tls_mgm.so"
modparam("tls_mgm","server_domain", "default")
modparam("tls_mgm","verify_cert", "[default]0")
modparam("tls_mgm","require_cert", "[default]0")
#modparam("tls_mgm","certificate", 
"[default]/etc/opensips/tls/rootCA/cacert.pem")
#modparam("tls_mgm","private_key", 
"[default]/etc/opensips/tls/rootCA/private/cakey.pem")
#modparam("tls_mgm","ca_list", 
"[bp3.erss.in]/etc/opensips/tls/user/user-calist.pem")
modparam("tls_mgm", "tls_method", "[default]SSLv23")
modparam("tls_mgm","tls_library","openssl")
#modparam("tls_mgm", "certificate", "[default]/etc/opensips/tls/user/caKey.pem")
#modparam("tls_mgm", "private_key", 
"[default]/etc/opensips/tls/user/privateKey.pem")
#modparam("tls_mgm", "client_domain", "14.139.
modparam("tls_mgm", "certificate", "[default]/home/ubuntu/cert.pem")
modparam("tls_mgm", "private_key", "[default]/home/ubuntu/privkey.pem")

loadmodule