Re: [OpenSIPS-Users] question on core statistics.

[Johan De Clercq]

Goed morning,

How can I then increase opensips’s internal queue size?

Best regards, Johan

Verzonden vanuit Outlook voor iOS<https://aka.ms/o0ukef>

Van: Bogdan-Andrei Iancu 
Verzonden: Thursday, April 18, 2024 4:29:41 PM
Aan: OpenSIPS users mailling list ; Johan De Clercq 

Onderwerp: Re: [OpenSIPS-Users] question on core statistics.

The `drop_requests` statistic is incremented when:
* the request is dropped by a pre-script callback (like B2B when there is no 
script execution for certain messages)
* the stateless `forward()` core function failed to send out something.

Regards,

Bogdan-Andrei Iancu

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

On 18.04.2024 17:19, Johan De Clercq wrote:
No I don't.
what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the default value of 
net.core.rmem_max and net.core.rmem_default.

Op do 18 apr 2024 om 16:02 schreef Ben Newlin 
mailto:ben.new...@genesys.com>>:

Are you calling drop() anywhere in your script?



https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13



Ben Newlin



From: Users 
mailto:users-boun...@lists.opensips.org>> on 
behalf of Johan De Clercq mailto:jo...@democon.be>>
Date: Thursday, April 18, 2024 at 5:27 AM
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] question on core statistics.

 EXTERNAL EMAIL - Please use caution with links and attachments





would it make sense to recompile with other flags ? And how do I set them  (I 
don't find these of menuconfig's compile options)?

Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.



Can somebody explain also what both flags mean.



Op do 18 apr 2024 om 11:07 schreef Johan De Clercq 
mailto:jo...@democon.be>>:

would it make sense to recompile with other flags ?

Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.



Can somebody explain also what both flags mean.





flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, 
FAST_LOCK-ADAPTIVE_WAIT

ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535

poll method support: poll, epoll, sigio_rt, select.



Op do 18 apr 2024 om 10:32 schreef Johan De Clercq 
mailto:jo...@democon.be>>:



Guys,



I have an opensips instance running with 24 worker children.

The worker load is very low.

UDP queues are on 50 megs.



when i query via the OS

cat /proc/net/udp

  sl  local_address rem_address   st tx_queue rx_queue tr tm->when retrnsmt   
uid  timeout inode ref pointer drops

  590: 03231D0A:13C4 : 07 : 00: 
 00 413684019 2 880074820bc0 0

  591: 03231D0A:13C5 : 07 : 00: 
 00 413766438 2 880465e4a440 0

  592: 03231D0A:13C6 : 07 : 00: 
 00 412035865 2 8803e5a56b80 0

  934: 01231D0A:151C : 07 : 00: 
 00 26790 2 88046c054840 0

  935: 0201FFEF:151D : 07 : 00: 
 00 26787 2 88046c054bc0 0

  935: 01231D0A:151D : 07 : 00: 
 00 26791 2 88046c0544c0 0

 1972: :D92A : 07 : 00: 
 00 15506 2 88046dce5040 0

 5479: :E6DD : 07 : 00: 
 00 22811 2 880465e4ab40 0

12075: AA0914AC:00A1 : 07 : 00: 
 00 20572 2 88086d020800 0

12075: 017F:00A1 : 07 : 00: 
 00 20571 2 88086d020b80 0

13320: :857E : 07 : 00:    
1000 17515 2 8800368ac780 0

15661: :CEA3 : 07 : 00: 
 00 15505 2 8800368acb00 0



=> no drops



what worries me is that there are drop requests and they  go up when I query 
via the mi interface

opensipsctl fifo get_statistics drop_requests

core:drop_requests:: 198107

opensipsctl fifo get_statistics drop_requests

core:drop_requests:: 199157

opensipsctl_reg fifo get_statistics drop_requests

core:drop_requests:: 204116



I don't see any memory issue, also the processload is low.





so 3 questions:

- what exactly is drop_request.

- do I need to worry about this

- how can I make them go lower.

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Re: [OpenSIPS-Users] question on core statistics.

[Johan De Clercq]

Wadii,

this is the beginning of route[0]

route {
if (!mf_process_maxfwd_header("70") && $retcode==-1)
{
sl_send_reply("483","Too Many Hops");
xlog("callid [$ci] from [$fU] to [$tU] loop detected");
exit;
};
force_rport();
t_on_failure("Trunk_On_Error");

if (has_totag()) {
loose_route();
t_relay();
exit;
}
record_route();
create_dialog("B");

so I don't think that this is related ?

Op do 18 apr 2024 om 16:12 schreef Wadii ELMAJDI | Evenmedia <
wa...@evenmedia.fr>:

> Calling exit() during the initial request and right before creating the
> dialog also increments the drop_requests statistic
>
>
>
> *De :* Users  *De la part de* Ben Newlin
> *Envoyé :* jeudi 18 avril 2024 15:59
> *À :* OpenSIPS users mailling list 
> *Objet :* Re: [OpenSIPS-Users] question on core statistics.
>
>
>
> Are you calling drop() anywhere in your script?
>
>
>
> https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13
>
>
>
> Ben Newlin
>
>
>
> *From: *Users  on behalf of Johan De
> Clercq 
> *Date: *Thursday, April 18, 2024 at 5:27 AM
> *To: *OpenSIPS users mailling list 
> *Subject: *Re: [OpenSIPS-Users] question on core statistics.
>
> * EXTERNAL EMAIL - Please use caution with links and attachments *
>
>
> --
>
> would it make sense to recompile with other flags ? And how do I set them
> (I don't find these of menuconfig's compile options)?
>
> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.
>
>
>
> Can somebody explain also what both flags mean.
>
>
>
> Op do 18 apr 2024 om 11:07 schreef Johan De Clercq :
>
> would it make sense to recompile with other flags ?
>
> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.
>
>
>
> Can somebody explain also what both flags mean.
>
>
>
>
>
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535
>
> poll method support: poll, epoll, sigio_rt, select.
>
>
>
> Op do 18 apr 2024 om 10:32 schreef Johan De Clercq :
>
>
>
> Guys,
>
>
>
> I have an opensips instance running with 24 worker children.
>
> The worker load is very low.
>
> UDP queues are on 50 megs.
>
>
>
> when i query via the OS
>
> cat /proc/net/udp
>
>   sl  local_address rem_address   st tx_queue rx_queue tr tm->when
> retrnsmt   uid  timeout inode ref pointer drops
>
>   590: 03231D0A:13C4 : 07 : 00:
>  00 413684019 2 880074820bc0 0
>
>   591: 03231D0A:13C5 : 07 : 00:
>  00 413766438 2 880465e4a440 0
>
>   592: 03231D0A:13C6 : 07 : 00:
>  00 412035865 2 8803e5a56b80 0
>
>   934: 01231D0A:151C : 07 : 00:
>  00 26790 2 88046c054840 0
>
>   935: 0201FFEF:151D : 07 : 00:
>  00 26787 2 88046c054bc0 0
>
>   935: 01231D0A:151D : 07 : 00:
>  00 26791 2 88046c0544c0 0
>
>  1972: :D92A : 07 : 00:
>  00 15506 2 88046dce5040 0
>
>  5479: :E6DD : 07 : 00:
>  00 22811 2 880465e4ab40 0
>
> 12075: AA0914AC:00A1 : 07 : 00:
>  00 20572 2 88086d020800 0
>
> 12075: 017F:00A1 : 07 : 00:
>  00 20571 2 88086d020b80 0
>
> 13320: :857E : 07 : 00:
>    1000 17515 2 8800368ac780 0
>
> 15661: :CEA3 : 07 : 00:
>  00 15505 2 8800368acb00 0
>
>
>
> => no drops
>
>
>
> what worries me is that there are drop requests and they  go up when I
> query via the mi interface
>
> opensipsctl fifo get_statistics drop_requests
>
> core:drop_requests:: 198107
>
> opensipsctl fifo get_statistics drop_requests
>
> core:drop_requests:: 199157
>
> opensipsctl_reg fifo get_statistics drop_requests
>
> core:drop_requests:: 204116
>
>
>
> I don't see any memory issue, also the processload is low.
>
>
>
>
>
> so 3 questions:
>
> - what exactly is drop_request.
>
> - do I need to worry about this
>
> - how can I make them go lower.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
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Re: [OpenSIPS-Users] question on core statistics.

[Johan De Clercq]

No I don't.
what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the default
value of net.core.rmem_max and net.core.rmem_default.

Op do 18 apr 2024 om 16:02 schreef Ben Newlin :

> Are you calling drop() anywhere in your script?
>
>
>
> https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13
>
>
>
> Ben Newlin
>
>
>
> *From: *Users  on behalf of Johan De
> Clercq 
> *Date: *Thursday, April 18, 2024 at 5:27 AM
> *To: *OpenSIPS users mailling list 
> *Subject: *Re: [OpenSIPS-Users] question on core statistics.
>
> * EXTERNAL EMAIL - Please use caution with links and attachments *
>
>
> --
>
> would it make sense to recompile with other flags ? And how do I set them
> (I don't find these of menuconfig's compile options)?
>
> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.
>
>
>
> Can somebody explain also what both flags mean.
>
>
>
> Op do 18 apr 2024 om 11:07 schreef Johan De Clercq :
>
> would it make sense to recompile with other flags ?
>
> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.
>
>
>
> Can somebody explain also what both flags mean.
>
>
>
>
>
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535
>
> poll method support: poll, epoll, sigio_rt, select.
>
>
>
> Op do 18 apr 2024 om 10:32 schreef Johan De Clercq :
>
>
>
> Guys,
>
>
>
> I have an opensips instance running with 24 worker children.
>
> The worker load is very low.
>
> UDP queues are on 50 megs.
>
>
>
> when i query via the OS
>
> cat /proc/net/udp
>
>   sl  local_address rem_address   st tx_queue rx_queue tr tm->when
> retrnsmt   uid  timeout inode ref pointer drops
>
>   590: 03231D0A:13C4 : 07 : 00:
>  00 413684019 2 880074820bc0 0
>
>   591: 03231D0A:13C5 : 07 : 00:
>  00 413766438 2 880465e4a440 0
>
>   592: 03231D0A:13C6 : 07 : 00:
>  00 412035865 2 8803e5a56b80 0
>
>   934: 01231D0A:151C : 07 : 00:
>  00 26790 2 88046c054840 0
>
>   935: 0201FFEF:151D : 07 : 00:
>  00 26787 2 88046c054bc0 0
>
>   935: 01231D0A:151D : 07 : 00:
>  00 26791 2 88046c0544c0 0
>
>  1972: :D92A : 07 : 00:
>  00 15506 2 88046dce5040 0
>
>  5479: :E6DD : 07 : 00:
>  00 22811 2 880465e4ab40 0
>
> 12075: AA0914AC:00A1 : 07 : 00:
>  00 20572 2 88086d020800 0
>
> 12075: 017F:00A1 : 07 : 00:
>  00 20571 2 88086d020b80 0
>
> 13320: :857E : 07 : 00:
>    1000 17515 2 8800368ac780 0
>
> 15661: :CEA3 : 07 : 00:
>  00 15505 2 8800368acb00 0
>
>
>
> => no drops
>
>
>
> what worries me is that there are drop requests and they  go up when I
> query via the mi interface
>
> opensipsctl fifo get_statistics drop_requests
>
> core:drop_requests:: 198107
>
> opensipsctl fifo get_statistics drop_requests
>
> core:drop_requests:: 199157
>
> opensipsctl_reg fifo get_statistics drop_requests
>
> core:drop_requests:: 204116
>
>
>
> I don't see any memory issue, also the processload is low.
>
>
>
>
>
> so 3 questions:
>
> - what exactly is drop_request.
>
> - do I need to worry about this
>
> - how can I make them go lower.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
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Re: [OpenSIPS-Users] question on core statistics.

[Johan De Clercq]

would it make sense to recompile with other flags ? And how do I set them
(I don't find these of menuconfig's compile options)?
Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.

Can somebody explain also what both flags mean.

Op do 18 apr 2024 om 11:07 schreef Johan De Clercq :

> would it make sense to recompile with other flags ?
> Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.
>
> Can somebody explain also what both flags mean.
>
>
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535
> poll method support: poll, epoll, sigio_rt, select.
>
> Op do 18 apr 2024 om 10:32 schreef Johan De Clercq :
>
>>
>> Guys,
>>
>> I have an opensips instance running with 24 worker children.
>> The worker load is very low.
>> UDP queues are on 50 megs.
>>
>> when i query via the OS
>> cat /proc/net/udp
>>   sl  local_address rem_address   st tx_queue rx_queue tr tm->when
>> retrnsmt   uid  timeout inode ref pointer drops
>>   590: 03231D0A:13C4 : 07 : 00:
>>  00 413684019 2 880074820bc0 0
>>   591: 03231D0A:13C5 : 07 : 00:
>>  00 413766438 2 880465e4a440 0
>>   592: 03231D0A:13C6 : 07 : 00:
>>  00 412035865 2 8803e5a56b80 0
>>   934: 01231D0A:151C : 07 : 00:
>>  00 26790 2 88046c054840 0
>>   935: 0201FFEF:151D : 07 : 00:
>>  00 26787 2 88046c054bc0 0
>>   935: 01231D0A:151D : 07 : 00:
>>  00 26791 2 88046c0544c0 0
>>  1972: :D92A : 07 : 00:
>>  00 15506 2 88046dce5040 0
>>  5479: :E6DD : 07 : 00:
>>  00 22811 2 880465e4ab40 0
>> 12075: AA0914AC:00A1 : 07 : 00:
>>  00 20572 2 88086d020800 0
>> 12075: 017F:00A1 : 07 : 00:
>>  00 20571 2 88086d020b80 0
>> 13320: :857E : 07 : 00:
>>    1000 17515 2 8800368ac780 0
>> 15661: :CEA3 : 07 : 00:
>>  00 15505 2 8800368acb00 0
>>
>> => no drops
>>
>> what worries me is that there are drop requests and they  go up when I
>> query via the mi interface
>> opensipsctl fifo get_statistics drop_requests
>> core:drop_requests:: 198107
>> opensipsctl fifo get_statistics drop_requests
>> core:drop_requests:: 199157
>> opensipsctl_reg fifo get_statistics drop_requests
>> core:drop_requests:: 204116
>>
>> I don't see any memory issue, also the processload is low.
>>
>>
>> so 3 questions:
>> - what exactly is drop_request.
>> - do I need to worry about this
>> - how can I make them go lower.
>>
>
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Re: [OpenSIPS-Users] question on core statistics.

[Johan De Clercq]

would it make sense to recompile with other flags ?
Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.

Can somebody explain also what both flags mean.


flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.

Op do 18 apr 2024 om 10:32 schreef Johan De Clercq :

>
> Guys,
>
> I have an opensips instance running with 24 worker children.
> The worker load is very low.
> UDP queues are on 50 megs.
>
> when i query via the OS
> cat /proc/net/udp
>   sl  local_address rem_address   st tx_queue rx_queue tr tm->when
> retrnsmt   uid  timeout inode ref pointer drops
>   590: 03231D0A:13C4 : 07 : 00:
>  00 413684019 2 880074820bc0 0
>   591: 03231D0A:13C5 : 07 : 00:
>  00 413766438 2 880465e4a440 0
>   592: 03231D0A:13C6 : 07 : 00:
>  00 412035865 2 8803e5a56b80 0
>   934: 01231D0A:151C : 07 : 00:
>  00 26790 2 88046c054840 0
>   935: 0201FFEF:151D : 07 : 00:
>  00 26787 2 88046c054bc0 0
>   935: 01231D0A:151D : 07 : 00:
>  00 26791 2 88046c0544c0 0
>  1972: :D92A : 07 : 00:
>  00 15506 2 88046dce5040 0
>  5479: :E6DD : 07 : 00:
>  00 22811 2 880465e4ab40 0
> 12075: AA0914AC:00A1 : 07 : 00:
>  00 20572 2 88086d020800 0
> 12075: 017F:00A1 : 07 : 00:
>  00 20571 2 88086d020b80 0
> 13320: :857E : 07 : 00:
>    1000 17515 2 8800368ac780 0
> 15661: :CEA3 : 07 : 00:
>  00 15505 2 8800368acb00 0
>
> => no drops
>
> what worries me is that there are drop requests and they  go up when I
> query via the mi interface
> opensipsctl fifo get_statistics drop_requests
> core:drop_requests:: 198107
> opensipsctl fifo get_statistics drop_requests
> core:drop_requests:: 199157
> opensipsctl_reg fifo get_statistics drop_requests
> core:drop_requests:: 204116
>
> I don't see any memory issue, also the processload is low.
>
>
> so 3 questions:
> - what exactly is drop_request.
> - do I need to worry about this
> - how can I make them go lower.
>
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[OpenSIPS-Users] question on core statistics.

[Johan De Clercq]

Guys,

I have an opensips instance running with 24 worker children.
The worker load is very low.
UDP queues are on 50 megs.

when i query via the OS
cat /proc/net/udp
  sl  local_address rem_address   st tx_queue rx_queue tr tm->when
retrnsmt   uid  timeout inode ref pointer drops
  590: 03231D0A:13C4 : 07 : 00:
 00 413684019 2 880074820bc0 0
  591: 03231D0A:13C5 : 07 : 00:
 00 413766438 2 880465e4a440 0
  592: 03231D0A:13C6 : 07 : 00:
 00 412035865 2 8803e5a56b80 0
  934: 01231D0A:151C : 07 : 00:
 00 26790 2 88046c054840 0
  935: 0201FFEF:151D : 07 : 00:
 00 26787 2 88046c054bc0 0
  935: 01231D0A:151D : 07 : 00:
 00 26791 2 88046c0544c0 0
 1972: :D92A : 07 : 00:
 00 15506 2 88046dce5040 0
 5479: :E6DD : 07 : 00:
 00 22811 2 880465e4ab40 0
12075: AA0914AC:00A1 : 07 : 00:
 00 20572 2 88086d020800 0
12075: 017F:00A1 : 07 : 00:
 00 20571 2 88086d020b80 0
13320: :857E : 07 : 00:
   1000 17515 2 8800368ac780 0
15661: :CEA3 : 07 : 00:
 00 15505 2 8800368acb00 0

=> no drops

what worries me is that there are drop requests and they  go up when I
query via the mi interface
opensipsctl fifo get_statistics drop_requests
core:drop_requests:: 198107
opensipsctl fifo get_statistics drop_requests
core:drop_requests:: 199157
opensipsctl_reg fifo get_statistics drop_requests
core:drop_requests:: 204116

I don't see any memory issue, also the processload is low.


so 3 questions:
- what exactly is drop_request.
- do I need to worry about this
- how can I make them go lower.
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[OpenSIPS-Users] another question on sip format's.

[Johan De Clercq]

Can param's be added anywhere in a user part.

e.g.
sip:0123456789;gw=case106@1.2.3.4 //this is for sure valid.
sip:gw=case107;0123456789@1.2.3.4
sip:gw=case108;0123456789;gw=case108@1.2.3.4

the fist one is for sure valid, but what about the other two ?

BR,
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[OpenSIPS-Users] on from and to header.

[Johan De Clercq]

This is more a protocol question then anything else.

A sends a call B
INVITE B@5.6.7.8
from: ;tag=1
to: 
call-id: 1
..

B responds with from and to in e164 format
200 OK
from: ;tag=1
to: ; tag=2
call-id: 1

My gut feeling says to me that A will not like this and hence that B should
respong with A and B instead of +32A and +32B.
But can somebody explain me why or why not?

Best regards, Johan.
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Re: [OpenSIPS-Users] Load Balancer add destination as probing status

[Johan De Clercq]

Can you tell me how to disable a load balancer destination via mi ?

On Tue, 9 Apr 2024, 20:02 John Sliney,  wrote:

> Is there currently a way to add a destination to the load_balancer set
> with a status of disabled but with probing enabled, like the dispatcher
> table 'state' column allows?
>
> I have an osips working as  a SIP ingress/egress for Kubernetes traffic
> and as Asterisk Pods are created they are added to the load_balancer table
> as destinations, but OpenSIPS will route traffic to them before they're
> fully ready.
> I've attempted to automatically mark them as disabled with the mi
> lb_status command and let the probe enable them, but disabling via mi seems
> to disable probing as well, so they're never actually enabled.
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Re: [OpenSIPS-Users] memory fragmentation when calling dr_reload.

[Johan De Clercq]

In addtion, I have 24 children, so can I increase in some way only the
process that handles the fifo requests ?

Op wo 3 apr 2024 om 15:33 schreef Johan De Clercq :

> Hi,
>
> A client has a very big dynamic routing rule set. (dr_rules >= 2.1 gb ).
> When reloading the db in opensips (dr_reload), I see below error in the
> log
>
> ERROR:core:fm_malloc: not enough free pkg memory (268008864 bytes left),
> please increase the "-M" command line parameter!
>
> the -M parameter is now at 256. Should I increase this to 3000 to work
> aroun this issue ?
>
>
>
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[OpenSIPS-Users] memory fragmentation when calling dr_reload.

[Johan De Clercq]

Hi,

A client has a very big dynamic routing rule set. (dr_rules >= 2.1 gb ).
When reloading the db in opensips (dr_reload), I see below error in the log

ERROR:core:fm_malloc: not enough free pkg memory (268008864 bytes left),
please increase the "-M" command line parameter!

the -M parameter is now at 256. Should I increase this to 3000 to work
aroun this issue ?
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[OpenSIPS-Users] https://datatracker.ietf.org/doc/html/rfc3261#section-13.3.1.1

[Johan De Clercq]

Question,

I always believed that fr_inv_timer should trigger when an invite is not
finished in due time.
This seems however to contradict the link in the title

 If the UAS is not able to answer the invitation immediately, it can
   choose to indicate some kind of progress to the UAC (for example, an
   indication that a phone is ringing).  This is accomplished with a
   provisional response between 101 and 199.  These provisional
   responses establish early dialogs and therefore follow the procedures
   of Section 12.1.1
 in
addition to those of Section 8.2.6
.  A UAS
MAY
   send as many provisional responses as it likes.  Each of these MUST
   indicate the same dialog ID.  However, these will not be delivered
   reliably.

   If the UAS desires an extended period of time to answer the INVITE,
   it will need to ask for an "extension" in order to prevent proxies
   from canceling the transaction.  A proxy has the option of canceling
   a transaction when there is a gap of 3 minutes between responses in a
   transaction.  To prevent cancellation, the UAS MUST send a non-100
   provisional response at every minute, to handle the possibility of
   lost provisional responses.

  An INVITE transaction can go on for extended durations when the
  user is placed on hold, or when interworking with PSTN systems
  which allow communications to take place without answering the
  call.  The latter is common in Interactive Voice Response (IVR)
  systems.


Can somebody please comment on this ?

BR, Johan.
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[OpenSIPS-Users] dialog termination.

[Johan De Clercq]

Is there a way to configure dialog module so that it upon timeout it
flushes the dialog without sending bye ?
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Re: [OpenSIPS-Users] Staless mode in opensips

[Johan De Clercq]

I usually make a mix f.e. on OPTIONS, subscribe, notify I send 200 OK with
sl_send_reply.
INVITE handling I always do stateful.
Normally, the dependencies of a module are described in the module
documentation.
I hope this helps.

BR, Johan.

Op di 5 dec 2023 om 12:04 schreef :

> Hi opensips experts,
>
> We have always worked Opensips in a statefull mode and the first words in
> the routing module description are “OpenSIPS is basically *only* a
> transaction statefull proxy”.
>
> I’ve also seen that there was a stateless module “The SL module allows
> OpenSIPS to act as a stateless UA server and generate replies to SIP
> requests without keeping state” but is it possible to use functionality of
> Opensips like dialplan and dispatcher without keeping state of the dialog ?
>
> Thanks for your help
>
> On belhalf of my colleague Guillaume
>
> BR, Amel
>
>
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Re: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC

[Johan De Clercq]

I agree Giovanni. If the decision is made to go nsa (even not in first 
release), that should be taken into account. That’s why we need to scope.

Verzonden vanuit Outlook voor iOS<https://aka.ms/o0ukef>

Van: Giovanni Maruzzelli 
Verzonden: Wednesday, November 29, 2023 6:45:58 PM
Aan: Bogdan-Andrei Iancu 
CC: Johan De Clercq ; Giovanni Maruzzelli 
; wg-...@lists.opensips.org 
; OpenSIPS users mailling list 

Onderwerp: Re: [WG-IMS] Scope of IMS in OpenSIPS - RFC

Yes, actually there is a difference between 5g and 4g infrastructure, that 
actually often involve different interfacing from IMS to it, particularly pcscf 
and icscf, eg: the way they interact with hss and pcf/pcrf.
Problem is that 4g infrastructure is different from 5g. When they implement 
4g+5g, they implement actually both (so, no problem)

4g+5g is called NSA (not stand alone)

A pure 5g is SA (stand alone) and offer different interfaces from the ones 
provided by 4g.

In NSA you (IMS) can behave like it's pure 4g (you use 4g interfaces to do all 
things, even for the 5g part)

In SA not at all, you must interface to 5g

The main difference for what ims is concerned is pcf vs pcrf

Let's note that most private networks (enterprise, etc) will be SA

Most carriers will obviously be NSA



answered from mobile, please pardon terseness and typos,
-giovanni

On Wed, Nov 29, 2023, 18:25 Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:
Hi Johan,

The lowest point we should address in the whole IMS arch is the P-CSCF, so we 
are agnostic to the actual transport layer below us (like the xG stuff). Or am 
I saying here something wrong and there are some implications to the upper 
layers ?

Regards,

Bogdan-Andrei Iancu

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

On 29.11.2023 18:07, Johan De Clercq wrote:
In addition, the IMS should be able to handle 4G and 5G calls.
In my opinion, we should no longer about 2 and 3 G as they are being phased out 
everywhere.

wkr,

Op wo 29 nov 2023 om 16:39 schreef Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>>:
Hi Giovanni,

Thanks for the feedback here, a valuable one as usual :).

On the HSS, what you are saying aligns with the my own thoughts - that its 
functioning logic is somehow outside the our scope here, but we need to pay 
attention to the interfacing (DIAMETER or HTTP2.0).

Now, on the AS side - as I understand, it holds whatever custom logic the 
operator may have in routing and proving services (included VAS's). So to say, 
I see it as a highly programmable component. And if so, what we need to provide 
here is probably a very high level interface / API to allow call manipulation 
in a very abstract way... :-/ ??

Best Regards,

Bogdan-Andrei Iancu

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

On 29.11.2023 11:11, Giovanni Maruzzelli wrote:
First of all:
CONGRATULATIONS to the OpenSIPS community !!!
(I believe this is the first step of a long and satisfying journey)

On the topic:
in addition to the CSCF component, I would like to see efforts on the AS 
(Application Server) component of the IMS infrastructure.

The AS is probably way the simplest of it all, it will probably require the 
least modifications/additions to OpenSIPS.

But I would say AS will be crucial to a lot of people/use cases.

While for sure there will be a lot of cases for our community to build the 
voice/video complete IMS infrastructure on top of private 5G networks in 
enterprises and public administrations, I see as very much relevant also the 
use case of building infrastructure to provide additional third party services 
to big carriers, and to big carriers partners.

Also, AS is the correct and manageable way to provide additional services even 
if you build the core IMS infrastructure.

About HSS: this is the sancta sanctorum of a carrier/provider
Apart from the venerable fraunhofer java implementation, now we can count on 
the flexible java implementation in https://github.com/nickvsnetworking/pyhss 
with a lot of features, good performances, and actually built for production.

I would say better we concentrate on accessing the various different protocols 
of HSS (diameter/http2) from the various components (each component in IMS 
access HSS with a different interface with different vocabularies and actions.

MGCF/MGW, if needed, will be a natural extension of our CSCF/AS architecture.

Just my two cents, to keep the ball rolling,

Congratulation again,

-giovanni


On Tue, Nov 28, 2023 at 2:02 PM Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:
Hi all,

(disclaimer : cross lists posting is not a good practice - we will do this only 
to catch the attention and get momentum with this initial topic)

As a first step here, is to work out the scope of the IMS implementation in 
OpenSIPS. IMS is a vast concept, with SIP and non-SIP components, and we want 

Re: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC

[Johan De Clercq]

In addition, the IMS should be able to handle 4G and 5G calls.
In my opinion, we should no longer about 2 and 3 G as they are being phased
out everywhere.

wkr,

Op wo 29 nov 2023 om 16:39 schreef Bogdan-Andrei Iancu :

> Hi Giovanni,
>
> Thanks for the feedback here, a valuable one as usual :).
>
> On the HSS, what you are saying aligns with the my own thoughts - that its
> functioning logic is somehow outside the our scope here, but we need to pay
> attention to the interfacing (DIAMETER or HTTP2.0).
>
> Now, on the AS side - as I understand, it holds whatever custom logic the
> operator may have in routing and proving services (included VAS's). So to
> say, I see it as a highly programmable component. And if so, what we need
> to provide here is probably a very high level interface / API to allow call
> manipulation in a very abstract way... :-/ ??
>
> Best Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 29.11.2023 11:11, Giovanni Maruzzelli wrote:
>
> First of all:
> CONGRATULATIONS to the OpenSIPS community !!!
> (I believe this is the first step of a long and satisfying journey)
>
> On the topic:
> in addition to the CSCF component, I would like to see efforts on the AS
> (Application Server) component of the IMS infrastructure.
>
> The AS is probably way the simplest of it all, it will probably require
> the least modifications/additions to OpenSIPS.
>
> But I would say AS will be crucial to a lot of people/use cases.
>
> While for sure there will be a lot of cases for our community to build the
> voice/video complete IMS infrastructure on top of private 5G networks in
> enterprises and public administrations, I see as very much relevant also
> the use case of building infrastructure to provide additional third party
> services to big carriers, and to big carriers partners.
>
> Also, AS is the correct and manageable way to provide additional services
> even if you build the core IMS infrastructure.
>
> About HSS: this is the sancta sanctorum of a carrier/provider
> Apart from the venerable fraunhofer java implementation, now we can count
> on the flexible java implementation in
> https://github.com/nickvsnetworking/pyhss with a lot of features, good
> performances, and actually built for production.
>
> I would say better we concentrate on accessing the various different
> protocols of HSS (diameter/http2) from the various components (each
> component in IMS access HSS with a different interface with
> different vocabularies and actions.
>
> MGCF/MGW, if needed, will be a natural extension of our CSCF/AS
> architecture.
>
> Just my two cents, to keep the ball rolling,
>
> Congratulation again,
>
> -giovanni
>
>
> On Tue, Nov 28, 2023 at 2:02 PM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi all,
>>
>> (disclaimer : cross lists posting is not a good practice - we will do
>> this only to catch the attention and get momentum with this initial topic)
>>
>> As a first step here, is to work out the scope of the IMS implementation
>> in OpenSIPS. IMS is a vast concept, with SIP and non-SIP components, and we
>> want to understand and agree on which components of IMS may be subject of
>> work from the OpenSIPS perspective. For example, we do consider the CSCF as
>> a must here, but we may explore the HSS, AS, MGW or other components.
>>
>> From the OpenSIPS perspective, we look for IMS components which are SIP
>> related. At least as a starting point. So, the first obvious candidate is
>> the *Call Session Control Function (CSCF)*. And here we need to look
>> into and address the specific functionalities of each sub-component:
>> * P-CSCF
>> * I-CSCF
>> * S-CSCF
>>
>> Again, these are the pretty obvious components, still may look into and
>> evaluate (if of an interest of the OpenSIPS IMS implementation) areas as:
>> * HSS (from interconnection perspective)
>> * MGCF / MGW  (from interconnection perspective)
>> * SIP AS
>> * others ?
>>
>> Any feedback (with explanations and arguments) about what we should
>> consider for our IMS implementation is more the welcome. I set here just a
>> simple starting point, with no limitations or so. Feel free to contribute
>> to the topic
>>
>>
>> Best regards,
>>
>> --
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>>   https://www.siphub.com
>>
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
>
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Re: [OpenSIPS-Users] Debug logs show To tag which are actually From tag

[Johan De Clercq]

I agree, but it's very confusing.

Op di 28 nov 2023 om 10:35 schreef Bogdan-Andrei Iancu :

> Hi Robert,
>
> As from SIP perspective both TO and FROM hdrs have the same syntax,
> internally OpenSIPS uses the same function "parse_to()" for parsing both
> hdrs. So, we are all good here :)
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 23.11.2023 21:36, Robert Dyck wrote:
>
> While running opensips in debug mode I noticed that for initial requests
> of dialog creating methods the debug logs were showing a To tag where none
> actually exists. The tag displayed was actually the From tag.
>
> INVITE sip:8@192.168.1.2 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.k7nLoQHpR;rport
> From: ;*tag=GSq6JzXiH *
> To: sip:8@192.168.1.2
> CSeq: 21 INVITE
> Call-ID: VCALnmUf8V
>
>
> Nov 23 11:02:32 [1131200] DBG:maxfwd:is_maxfwd_present: value = 70
> Nov 23 11:02:32 [1131200] *DBG:sipmsgops:has_totag: no totag *
> Nov 23 11:02:32 [1131200] Initial request, method is INVITE, URI is
> sip:8@192.168.1.2
>
> Nov 23 11:02:32 [1131200] DBG:core:check_self: host != me
> Nov 23 11:02:32 [1131200] DBG:core:parse_headers: flags=78
> Nov 23 11:02:32 [1131200] DBG:tm:t_lookup_request: start searching:
> hash=42361, isACK=0
> Nov 23 11:02:32 [1131200] DBG:tm:matching_3261: RFC3261 transaction
> matching failed
> Nov 23 11:02:32 [1131200] DBG:tm:t_lookup_request: no transaction found
> Nov 23 11:02:32 [1131200] *DBG:core:parse_to_param: tag=GSq6JzXiH *
> Nov 23 11:02:32 [1131200] DBG:core:parse_to_param: end of header reached,
> state=11
>
>
> PUBLISH sip:7@192.168.1.2 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.myBdBZt3P;rport
> From: ;*tag=cbp~iz4fq*
> To: sip:7@192.168.1.2
> CSeq: 21 PUBLISH
> Call-ID: cRC6Gw590w
> Max-Forwards: 70
>
>
>
> Nov 23 11:07:08 [1133909] DBG:maxfwd:is_maxfwd_present: value = 70
> Nov 23 11:07:08 [1133909] *DBG:sipmsgops:has_totag: no totag*
> Nov 23 11:07:08 [1133909] Initial request, method is PUBLISH, URI is
> sip:7@192.168.1.2
>
>
>
> Nov 23 11:07:08 [1133909] DBG:core:check_self: host != me
> Nov 23 11:07:08 [1133909] DBG:core:parse_headers: flags=78
> Nov 23 11:07:08 [1133909] DBG:tm:t_lookup_request: start searching:
> hash=7222, isACK=0
> Nov 23 11:07:08 [1133909] DBG:tm:matching_3261: RFC3261 transaction
> matching failed
> Nov 23 11:07:08 [1133909] DBG:tm:t_lookup_request: no transaction found
> Nov 23 11:07:08 [1133909] *DBG:core:parse_to_param: tag=cbp~iz4fq*
> Nov 23 11:07:08 [1133909] DBG:core:parse_to_param: end of header reached,
> state=11
>
>
>
> SUBSCRIBE sip:rls@192.168.1.2 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.OkkDHBafu;rport
> From: ;*tag=g2aNdkn4o *
> To: sips:rls@192.168.1.2
> CSeq: 21 SUBSCRIBE
> Call-ID: L9LKmnZVY7
> Max-Forwards: 69
>
>
> Nov 23 11:07:07 [1133909] DBG:maxfwd:is_maxfwd_present: value = 70
> Nov 23 11:07:07 [1133909] *DBG:sipmsgops:has_totag: no totag*
> Nov 23 11:07:07 [1133909] Initial request, method is SUBSCRIBE, URI is
> sip:rls@192.168.1.2
>
>
>
> Nov 23 11:07:07 [1133909] DBG:core:check_self: host != me
> Nov 23 11:07:07 [1133909] DBG:core:parse_headers: flags=78
> Nov 23 11:07:07 [1133909] DBG:tm:t_lookup_request: start searching:
> hash=33557, isACK=0
> Nov 23 11:07:07 [1133909] DBG:tm:matching_3261: RFC3261 transaction
> matching failed
> Nov 23 11:07:07 [1133909] DBG:tm:t_lookup_request: no transaction found
> Nov 23 11:07:07 [1133909] *DBG:core:parse_to_param: tag=g2aNdkn4o *
> Nov 23 11:07:07 [1133909] DBG:core:parse_to_param: end of header reached,
> state=1
>
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Re: [OpenSIPS-Users] dlg_end_dlg in early state, how to rewrite 487 to 404

[Johan De Clercq]

forward to a self defined route and change it there.

route[404to487]
{


}
onreply_route
{

route(404to487);
}

Op wo 15 nov 2023 om 12:04 schreef Alain Bieuzent :

> Hi All,
>
>
>
> I have a case where I need to terminate an early state dialog but not with
> a 487 but by a 404.
>
>
>
> t_check_status() on onreply_route didn’t match, and it’s not allowed to
> use t_check_status() on local_route
>
>
>
> what is the best way to do it ?
>
>
>
> thanks
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Re: [OpenSIPS-Users] [OpenSIPS-Business] Introducing OpenSIPS 3.5

[Johan De Clercq]

Me too.


On Tue, 14 Nov 2023, 19:02 Giovanni Maruzzelli,  wrote:

> On Tue, Nov 14, 2023 at 3:34 PM Bogdan-Andrei Iancu 
> wrote:
>
>>
>> This year we introduce a new concept of an *OpenSIPS Working Group*. And
>> the *IMS OpenSIPS Working Group*
>>  is the
>> first one, aiming to gather people with inters in IMS with the goal to
>> draft, design and implement the IMS support in OpenSIPS.
>>
>> More details on this may be found here, please read and act:
>>
>> https://www.opensips.org/Development/Opensips-3-5-Planning
>>
>>
>
>
> G R E A T !!
> (count me on)
>
>
>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
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Re: [OpenSIPS-Users] opensips summit

[Johan De Clercq]

good plan.  I haven't visited munich in ages.

Op di 7 nov 2023 om 15:10 schreef Stefan Hofmeir :

> Hi,
>
> perhaps Munich would be also a great city for the next OpenSIPS Summit.
> As a native of Munich I could organize recommending hotels or the social
> event.
>
> --
> BRs
> Stefan
>
>
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[OpenSIPS-Users] opensips summit

[Johan De Clercq]

list,
do we have somebody who can arrange something in Vienna ?
looks like a great location to me :-)
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Re: [OpenSIPS-Users] t_new_request and X-headers

[Johan De Clercq]

write the info to a custom table and fetch it from there ?


Op ma 9 okt 2023 om 16:03 schreef Mickael Hubert :

> Hi all,
> I need to send an INFO message with X-headers from other dialog.
> I use t_new_request to send my request (apart from first dialog): OK
> I use local_route to add fixed X-header: OK
> But I want to copy some information from the first dialog to the new
> dialog (created by t_new_request). How can I do that please ?
>
> Thanks in advance
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Re: [OpenSIPS-Users] t_new_request and X-headers

[Johan De Clercq]

avp_dbquery if I am not wrong.

Op ma 9 okt 2023 om 16:31 schreef Johan De Clercq :

> write the info to a custom table and fetch it from there ?
>
>
> Op ma 9 okt 2023 om 16:03 schreef Mickael Hubert :
>
>> Hi all,
>> I need to send an INFO message with X-headers from other dialog.
>> I use t_new_request to send my request (apart from first dialog): OK
>> I use local_route to add fixed X-header: OK
>> But I want to copy some information from the first dialog to the new
>> dialog (created by t_new_request). How can I do that please ?
>>
>> Thanks in advance
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>>
>
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Re: [OpenSIPS-Users] on routing invite's in a trunk context.

[Johan De Clercq]

It’s a very non rfc scenario. Source ip and via are the ip of a loadbalancer 
that I need to skip …. ;-(. The contact does have the correct ip. Anyway, you 
gave me a good hint. I will add the contact ip as record route and see what 
that gives.

Verzonden vanuit Outlook voor iOS

Van: Users  namens Ben Newlin 

Verzonden: Friday, October 6, 2023 5:35:40 PM
Aan: OpenSIPS users mailling list 
Onderwerp: Re: [OpenSIPS-Users] on routing invite's in a trunk context.


Johan,



This is actually a pretty standard SIP flow that we use all the time. I 
recommend this article [1] for an overview of how routing works in SIP.



In short, the Contact and Record-Route headers are only used for routing of 
requests not responses, and mostly only sequential requests which are requests 
within a dialog. The topology_hiding_match and loose_route functions both 
operate on requests only, so they can both only be called from a request_route 
[1] [2].



The 200 OK is a response within the initial INVITE transaction. Responses 
within a SIP transaction are routed based on the source IP of request and/or 
the Via headers. OpenSIPS should just handle that for you, unless you have some 
strange routing.



[1] 
https://kb.smartvox.co.uk/opensips/contact-and-record-route-headers-explained/

[2] 
https://opensips.org/docs/modules/3.2.x/topology_hiding.html#func_topology_hiding_match

[3] https://opensips.org/docs/modules/3.2.x/rr.html#func_loose_route



Ben Newlin



From: Users  on behalf of johan 

Date: Friday, October 6, 2023 at 10:30 AM
To: OpenSIPS users mailling list 
Subject: [OpenSIPS-Users] on routing invite's in a trunk context.

 EXTERNAL EMAIL - Please use caution with links and attachments

This is a general question on routing (to be honest: it is a really
strange case).


A has a sip trunk to B (opensips), B has a sip trunk to C



A sends an invite to B with contact header A'   and record-route header
to A''

B string the record_route header , calls topology hiding and droutes the
call to C

C sends 200 OK back but it needs to be routed on the content of the
contact header.


Now how does this route to A ?   Is this the default
topology_hiding_match() case ? Or do I need to do something special ?






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Re: [OpenSIPS-Users] Can't set TLS ciphers_list to NULL

[Johan De Clercq]

I find it strange to start with the null suite. What version of this is
this?

On Mon, 2 Oct 2023, 11:36 Răzvan Crainea,  wrote:

> Hi, Matt!
>
> Are you sure that wolfssl supports the NULL cipher list? You can see all
> the available ciphers when OpenSIPS starts. For example, my setup has
> the following ciphers:
>
> ```
> Oct  2 09:56:43 [207525] INFO:tls_wolfssl:_wolfssl_show_ciphers:
> Ciphers:
>
> TLS13-AES128-GCM-SHA256:TLS13-AES256-GCM-SHA384:TLS13-CHACHA20-POLY1305-SHA256:TLS13-AES128-CCM-SHA256:TLS13-AES128-CCM-8-SHA256:TLS13-AES128-CCM8-SHA256:TLS13-SHA256-SHA256:TLS13-SHA384-SHA384:RC4-SHA:RC4-MD5:DES-CBC3-SHA:AES128-SHA:AES256-SHA:NULL-MD5:NULL-SHA:NULL-SHA256:DHE-RSA-AES128-SHA:DHE-RSA-AES256-SHA:DHE-PSK-AES256-GCM-SHA384:DHE-PSK-AES128-GCM-SHA256:DHE-PSK-AES256-CBC-SHA384:DHE-PSK-AES128-CBC-SHA256:DHE-PSK-AES128-CCM:DHE-PSK-AES256-CCM:DHE-PSK-NULL-SHA384:DHE-PSK-NULL-SHA256:AES128-CCM-8:AES128-CCM8:AES256-CCM-8:AES256-CCM8:ECDHE-ECDSA-AES128-CCM:ECDHE-ECDSA-AES128-CCM-8:ECDHE-ECDSA-AES128-CCM8:ECDHE-ECDSA-AES256-CCM-8:ECDHE-ECDSA-AES256-CCM8:ECDHE-RSA-AES128-SHA:ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES128-SHA:ECDHE-ECDSA-AES256-SHA:ECDHE-RSA-RC4-SHA:ECDHE-RSA-DES-CBC3-SHA:ECDHE-ECDSA-RC4-SHA:ECDHE-ECDSA-DES-CBC3-SHA:AES128-SHA256:AES256-SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA256:AES128-GCM-SHA256:AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES256-GCM-SHA384:CAMELLIA128-SHA:DHE-RSA-CAMELLIA128-SHA:CAMELLIA256-SHA:DHE-RSA-CAMELLIA256-SHA:CAMELLIA128-SHA256:DHE-RSA-CAMELLIA128-SHA256:CAMELLIA256-SHA256:DHE-RSA-CAMELLIA256-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:ECDHE-RSA-CHACHA20-POLY1305:ECDHE-ECDSA-CHACHA20-POLY1305:DHE-RSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305-OLD:ECDHE-ECDSA-CHACHA20-POLY1305-OLD:DHE-RSA-CHACHA20-POLY1305-OLD:ADH-AES128-SHA:ADH-AES256-GCM-SHA384:ECDHE-ECDSA-NULL-SHA:ECDHE-PSK-NULL-SHA256:ECDHE-PSK-AES128-CBC-SHA256:ECDHE-PSK-AES128-GCM-SHA256:PSK-CHACHA20-POLY1305:ECDHE-PSK-CHACHA20-POLY1305:DHE-PSK-CHACHA20-POLY1305:EDH-RSA-DES-CBC3-SHA:WDM-NULL-SHA256
>
> ```
>
> And plain NULL cipher is not available, only a set of its other variants.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer / SIPhub CTO
> http://www.opensips-solutions.com / https://www.siphub.com
>
> On 9/30/23 17:16, L S wrote:
> > Wolfssl gives an error and Opensips doesn't start when trying to set the
> > ciphers_list to NULL for a client domain in 3.2.13.
> >
> > modparam("tls_mgm", "ciphers_list", "[testclient]NULL")
> >
> > ERROR:tls_wolfssl:_wolfssl_init_tls_dom: failure to set SSL context
> > cipher list 'NULL'
> >
> > Any suggestions?
> >
> > Thanks,
> > Matt
> >
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Re: [OpenSIPS-Users] Configuration file

[Johan De Clercq]

Indeed. Everything starts with reading documentation.

On Thu, 14 Sept 2023, 20:23 Callum Guy,  wrote:

> The idea is that you create your own based on your unique
> infrastructure requirements.
>
> Learn you must.
>
> https://www.opensips.org/Documentation/Manual-3-3
>
>
> On Thu, 14 Sept 2023 at 15:21, Prathibha B 
> wrote:
> >
> > I need the conf file for opensips 3.3
> >
> > Sent from Outlook for Android
> > 
> > From: Prathibha B 
> > Sent: Thursday, September 14, 2023 6:23:35 PM
> > To: users@lists.opensips.org 
> > Subject: Configuration file
> >
> > Can anyone share the opensips conf file with rtpengine?
> >
> > Sent from Outlook for Android
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
>
>
>
>
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Re: [OpenSIPS-Users] How to fetch data from OpenSIPS using external app

[Johan De Clercq]

Get it from db. For the register, use an open source softphone or sipsak or
sipp.

On Fri, Nov 11, 2022, 09:13 cheny via Users 
wrote:

> Hi everyone,
>
> I need to write a program to mean below functions:
>
> 1.register account to opensips.
> 2.fetch registered users from opensips.
> 3.fetch ongoing calls from opensips.
> 4.fetch history calls from opensips, acc moudle.
> According opensips "Management Interface (MI)" Overview, some MI commands
> allows an external app to push data into OpenSIPS(like registering a
> contact), or fetch data from OpenSIPS(like see registered users, see
> ongoing calls).
>
> When I look into almost all the core and module guide, I can't find any MI
> command to support functions that I want.
>
> Did I miss something?  What would be the best way to handle it?
>
> Any hint will be very helpful!
>
>
> Best regards.
>
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Re: [OpenSIPS-Users] The update from yesterday makes all calls fail after 20 seconds, how do I go back?

[Johan De Clercq]

Xlog(….);

Outlook voor iOS downloaden

Van: Users  namens Saint Michael 

Verzonden: Wednesday, September 14, 2022 9:56:41 PM
Aan: OpenSIPS users mailling list 
Onderwerp: Re: [OpenSIPS-Users] The update from yesterday makes all calls fail 
after 20 seconds, how do I go back?

how do I do this:
" put some log on local_route"
Sorry I am learning


On Wed, Sep 14, 2022 at 3:55 PM Daniel Zanutti 
mailto:daniel.zanu...@gmail.com>> wrote:
So your Opensips is hanging up the call.

Do you see any log on it? Try put some log on local_route if you don't see 
anything.



On Wed, Sep 14, 2022 at 4:40 PM Saint Michael 
mailto:vene...@gmail.com>> wrote:
This is a trace showing a BYE from Opensips, but none of the sides did actually 
hangup.


On Wed, Sep 14, 2022 at 3:33 PM Saint Michael 
mailto:vene...@gmail.com>> wrote:
I use opensips 3.1, and I did an update yesterday. in all the boxes that I 
upgraded all calls fail after 20 seconds.
cd /usr/src/opensips-3.1/
git pull
make clean;make proper;make all
make modules
make install
clearlog.sh
systemctl restart opensips
opensips -V



How do I go back?


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Re: [OpenSIPS-Users] opensips-cli debian 11

[Johan De Clercq]

Debian 11

Outlook voor iOS downloaden

Van: Users  namens Răzvan Crainea 

Verzonden: Wednesday, August 24, 2022 1:17:00 PM
Aan: users@lists.opensips.org 
Onderwerp: Re: [OpenSIPS-Users] opensips-cli debian 11

Hi, Johan!

Are you trying to install on Debian 11 or on Ubuntu 22?
Because I see that the sources list is jammy, but then you are fetching
bullseye.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 8/24/22 13:42, johan wrote:
> the old repo gives errors on bullseye.
>
> root@sipp:/etc/apt/sources.list.d# apt install opensips-cli
> Reading package lists... Done
> Building dependency tree... Done
> Reading state information... Done
> The following additional packages will be installed:
>libpq5 python-babel-localedata python3-anyjson python3-arrow python3-babel
>python3-cffi python3-cffi-backend python3-cryptography python3-dateutil
>python3-infinity python3-mysqldb python3-openssl python3-ply
>python3-psycopg2cffi python3-pycparser python3-pymysql python3-sqlalchemy
>python3-sqlalchemy-ext python3-sqlalchemy-utils python3-tz
> Suggested packages:
>python-arrow-doc python-cryptography-doc python3-cryptography-vectors
>python3-mysqldb-dbg python-openssl-doc python3-openssl-dbg python-ply-doc
>python-pymysql-doc python-sqlalchemy-doc python3-fdb python3-pymssql
>python3-psycopg2 python-sqlalchemy-utils-doc
> The following NEW packages will be installed:
>libpq5 opensips-cli python-babel-localedata python3-anyjson python3-arrow
>python3-babel python3-cffi python3-cffi-backend python3-cryptography
>python3-dateutil python3-infinity python3-mysqldb python3-openssl
>python3-ply python3-psycopg2cffi python3-pycparser python3-pymysql
>python3-sqlalchemy python3-sqlalchemy-ext python3-sqlalchemy-utils
>python3-tz
> 0 upgraded, 21 newly installed, 0 to remove and 0 not upgraded.
> Need to get 7,133 kB of archives.
> After this operation, 38.5 MB of additional disk space will be used.
> Do you want to continue? [Y/n] y
> Get:1 http://deb.debian.org/debian bullseye/main amd64 libpq5 amd64
> 13.7-0+deb11u1 [180 kB]
> Get:2 http://deb.debian.org/debian bullseye/main amd64
> python3-sqlalchemy all 1.3.22+ds1-1 [795 kB]
> Get:3 https://apt.opensips.org jammy/cli-nightly amd64 opensips-cli all
> 0.1~20220822~a480e53-1 [41.3 kB]
> Get:4 http://deb.debian.org/debian bullseye/main amd64 python3-anyjson
> all 0.3.3-2 [8,196 B]
> Get:5 http://deb.debian.org/debian bullseye/main amd64 python3-dateutil
> all 2.8.1-6 [79.2 kB]
> Get:6 http://deb.debian.org/debian bullseye/main amd64 python3-arrow all
> 0.17.0-1 [50.7 kB]
> Get:7 http://deb.debian.org/debian bullseye/main amd64
> python-babel-localedata all 2.8.0+dfsg.1-7 [4,997 kB]
> Get:8 http://deb.debian.org/debian bullseye/main amd64 python3-tz all
> 2021.1-1 [34.8 kB]
> Get:9 http://deb.debian.org/debian bullseye/main amd64 python3-babel all
> 2.8.0+dfsg.1-7 [100 kB]
> Get:10 http://deb.debian.org/debian bullseye/main amd64 python3-infinity
> all 1.5-2 [4,364 B]
> Get:11 http://deb.debian.org/debian bullseye/main amd64
> python3-cffi-backend amd64 1.14.5-1 [85.8 kB]
> Get:12 http://deb.debian.org/debian bullseye/main amd64 python3-ply all
> 3.11-4 [65.5 kB]
> Get:13 http://deb.debian.org/debian bullseye/main amd64
> python3-pycparser all 2.20-3 [74.5 kB]
> Get:14 http://deb.debian.org/debian bullseye/main amd64 python3-cffi all
> 1.14.5-1 [87.9 kB]
> Get:15 http://deb.debian.org/debian bullseye/main amd64
> python3-psycopg2cffi amd64 2.8.1-2 [64.1 kB]
> Get:16 http://deb.debian.org/debian bullseye/main amd64
> python3-sqlalchemy-utils all 0.36.8-4 [66.6 kB]
> Get:17 http://deb.debian.org/debian bullseye/main amd64
> python3-cryptography amd64 3.3.2-1 [223 kB]
> Get:18 http://deb.debian.org/debian bullseye/main amd64 python3-openssl
> all 20.0.1-1 [53.7 kB]
> Get:19 http://deb.debian.org/debian bullseye/main amd64 python3-mysqldb
> amd64 1.4.4-2+b3 [57.0 kB]
> Get:20 http://deb.debian.org/debian bullseye/main amd64 python3-pymysql
> all 0.9.3-2 [43.4 kB]
> Get:21 http://deb.debian.org/debian bullseye/main amd64
> python3-sqlalchemy-ext amd64 1.3.22+ds1-1 [19.9 kB]
> Fetched 7,133 kB in 14s (519
> kB/s)
> Selecting previously unselected package libpq5:amd64.
> (Reading database ... 163371 files and directories currently installed.)
> Preparing to unpack .../00-libpq5_13.7-0+deb11u1_amd64.deb ...
> Unpacking libpq5:amd64 (13.7-0+deb11u1) ...
> Selecting previously unselected package python3-sqlalchemy.
> Preparing to unpack .../01-python3-sqlalchemy_1.3.22+ds1-1_all.deb ...
> Unpacking python3-sqlalchemy (1.3.22+ds1-1) ...
> Selecting previously unselected package python3-anyjson.
> Preparing to unpack .../02-python3-anyjson_0.3.3-2_all.deb ...
> Unpacking python3-anyjson (0.3.3-2) ...
> Selecting previously unselected package python3-dateutil.
> Preparing to unpack .../03-python3-dateutil_2.8.1-6_all.deb ...
> 

Re: [OpenSIPS-Users] Opensips active (exited)

[Johan De Clercq]

Check daemon path

Outlook voor iOS downloaden

Van: Users  namens Ali Alawi 

Verzonden: Thursday, May 19, 2022 9:30:50 PM
Aan: OpenSIPS users mailling list 
Onderwerp: [OpenSIPS-Users] Opensips active (exited)

Dear all,

I have installed opensips 3.2 using the git clone --recursive and compile it 
using make menuconfig, mysql (mariadb) and cli are all set correctly.

when I start opensips the status shows no error but only active (exited) as 
shown below:

systemctl status opensips
● opensips.service - LSB: Start the OpenSIPS SIP server
   Loaded: loaded (/etc/init.d/opensips; generated)
   Active: active (exited) since Thu 2022-05-19 14:30:15 EDT; 1min 49s ago
 Docs: man:systemd-sysv-generator(8)
  Process: 433 ExecStart=/etc/init.d/opensips start (code=exited, 
status=0/SUCCESS)

May 19 14:30:15 debsips systemd[1]: Starting LSB: Start the OpenSIPS SIP 
server...
May 19 14:30:15 debsips systemd[1]: Started LSB: Start the OpenSIPS SIP server.


I tried several ways of installation and configuration but the opensips doesn't 
start properly, I am not able to register clients (even though I have 
successfully added client and domain from CP). also there is no log recorded in 
opensips.log.

Please give me any advice on this matter. I want to see the status of active 
(running).

Regards,
Ali
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Re: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli?

[Johan De Clercq]

Create  your own table in the db.
Then specify that as DB table in the module parameters (will avoid hassle
when you have to migrate)

Op wo 18 mei 2022 om 08:22 schreef Bela H :

> Hello,
>
> I want to set up a call forwarding and followed the instructions from the
> "Building telephony systems with OpenSIPS". However, this is a little bit
> outdated, the old opensipsctl was replaced by opensips-cli. How can I add
> an avp data into a usr_preferences table from opensips-cli?
> This was the old format: *opensipsctl avp add A_number callfwd 0 C_number*
>
> Cheers,
> Bela
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Re: [OpenSIPS-Users] Maximum number of ReINVITEs option?

[Johan De Clercq]

Fail2ban if they come quickly?

Outlook voor iOS downloaden

Van: Users  namens Ray Jackson 

Verzonden: Saturday, May 14, 2022 4:42:07 AM
Aan: users@lists.opensips.org 
Onderwerp: [OpenSIPS-Users] Maximum number of ReINVITEs option?

Hi all,

I have a misbehaving SIP client which is repeatedly sending SIP ReINVITE
messages over and over again (in the hundreds) and want to enforce a
limit on the number of ReINVITE messages per Call to stop this in it's
tracks.  The CSeq is incrementing on each ReINVITE.

Are there any config settings to limit the number of ReINVITEs on a
single Call/Dialog?  I was going to possibly keep a count of the INVITEs
using a cache variable based on the Unique CallID but wanted to check
first there wasn't a simpler (built-in) way of doing this?

Thanks,
Ray




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Re: [OpenSIPS-Users] inject_dtmf

[Johan De Clercq]

I spoke with sipwise yesterday and according to them, the problem is that I
send the dtmf too early (immediately in reply route after
rtpengine_answer).
I will try this evening with setting up a call and then injecting the dtmf
with curl or something.

This leads me to the follow up question : how can I send a command from the
script after a certain time ?   I thought about timer-route, but there you
don't have the call id.

Op wo 23 mrt. 2022 om 09:20 schreef Răzvan Crainea :

> Hi, Johan!
>
> Can you post opensips logs of rtpengine module? Are there any errors.
> Also, what version of OpenSIPS are you using?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 3/22/22 13:34, Johan De Clercq wrote:
> > Hi,
> >
> > for one reason or another I don't get this working.
> > What I do
> > 1. when the invite is send, i call rtpengine_offer with inject_DTMF flag.
> > 2. in the onreply route, I call rtpengine_answer with inject_DTMF.  Then
> > I call rtpengine_playdtmf("0").
> >
> > The dtmf is NEVER send out.
> >
> > What do I do wrong here ?
> > Is there somebody with experience on this ?
> >
> > wkr,
> >
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[OpenSIPS-Users] inject_dtmf

[Johan De Clercq]

Hi,

for one reason or another I don't get this working.
What I do
1. when the invite is send, i call rtpengine_offer with inject_DTMF flag.
2. in the onreply route, I call rtpengine_answer with inject_DTMF.  Then I
call rtpengine_playdtmf("0").

The dtmf is NEVER send out.

What do I do wrong here ?
Is there somebody with experience on this ?

wkr,
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Re: [OpenSIPS-Users] Replacing From Number

[Johan De Clercq]

;-)

On Tue, Mar 8, 2022, 15:39 Alain Bieuzent  wrote:

> Hi Alex,
>
>
>
> This the day of rewriting from ….
>
>
>
> Try with uac_replace_from([display],uri)
>
>
>
> Regards
>
>
>
> *De : *Users  au nom de Alexander
> Perkins 
> *Répondre à : *OpenSIPS users mailling list 
> *Date : *mardi 8 mars 2022 à 15:30
> *À : *OpenSIPS users mailling list 
> *Objet : *[OpenSIPS-Users] Replacing From Number
>
>
>
> Hi All.  I am trying to replace the From Number to match that of our main
> office line, but I do not seem to be having luck with it.  Here's what I am
> doing:
>
>
>
> This is from the PCAP - From: 12121212  >;tag=85698821
>
>
>
> This is the part of my script I am using to try to replace it:
>
>
>
> replace_body_all('$fU','19905551212');
>
>
>
> However, the From Number is never replaced.  What should I do?
>
>
>
> Thank you,
>
> Alex
>
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Re: [OpenSIPS-Users] how to change mysql data directory to CP

[Johan De Clercq]

Do cd /
Locate db.inc.php

Outlook voor iOS downloaden

Van: Users  namens mrsanvicente 

Verzonden: Wednesday, December 29, 2021 2:51:07 AM
Aan: Bogdan-Andrei Iancu 
CC: OpenSIPS users mailling list 
Onderwerp: Re: [OpenSIPS-Users] how to change mysql data directory to CP

Hello Bogdan-Andrei / All,

Sorry for the delay.  I just didn’t find a path or var to modify in  db.inc.php.


But later after a reboot, opensips stop working because it could not find the 
new path to the db.

So the source of the problem might be,  how to change path to mysql to opensips.
 Thanks
Saludos
Mario San Vicente


El 17 dic 2021, a la(s) 9:16, Bogdan-Andrei Iancu  
escribió:

 You need to update the DB settings in this file:

https://github.com/OpenSIPS/opensips-cp/blob/master/config/db.inc.php

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS eBootcamp 2021
  https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 12/16/21 1:52 AM, mrsanvicente wrote:
Yes i did.   And restarted mysqld and httpd.  But still not working.


Thanks

Mario San Vicente


El 15 dic 2021, a la(s) 16:14, Joseph Jackson 
 escribió:



Did you also change the location of the socket file in the my.cnf?







From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Mario San 
Vicente
Sent: Wednesday, December 15, 2021 11:27 AM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] how to change mysql data directory to CP



Hello Everyone,



I have changed my mysql directory and a had limited space on the default 
partition.   I applied the change and opensips is working fine.  But the 
control panel is not working and i can not find where to update it.



Getting the following error:





http://x.x.x.x/cp/login.php

Error!: SQLSTATE[HY000] [2002] Can't connect to local MySQL server through 
socket '/var/lib/mysql/mysql.sock' (2)



Any idea, where to update this?



Saludos!





Mario San Vicente



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Re: [OpenSIPS-Users] timers in opensips 3.2.

[Johan De Clercq]

Indeed.

Outlook voor iOS downloaden

Van: Bogdan-Andrei Iancu 
Verzonden: Friday, December 17, 2021 4:14:33 PM
Aan: OpenSIPS users mailling list ; johan 

Onderwerp: Re: timers in opensips 3.2.

Hi Johan,

I assume you refer to the fr_timeout, right ?

https://opensips.org/html/docs/modules/3.2.x/tm.html#param_fr_timeout

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS eBootcamp 2021
  https://opensips.org/training/OpenSIPS_eBootcamp_2021/

On 12/17/21 1:01 PM, johan wrote:

Can it be that the retransmission timer now defautls to 10 ms ?

If yes, can you then please update the documentation ?


wkr,





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Re: [OpenSIPS-Users] how to change mysql data directory to CP

[Johan De Clercq]

Isn't there a ...db.inc.php?

On Thu, Dec 16, 2021, 00:56 mrsanvicente  wrote:

> Yes i did.   And restarted mysqld and httpd.  But still not working.
>
>
> Thanks
>
> Mario San Vicente
>
>
> El 15 dic 2021, a la(s) 16:14, Joseph Jackson 
> escribió:
>
> 
>
> Did you also change the location of the socket file in the my.cnf?
>
>
>
>
>
>
>
> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behalf Of *Mario
> San Vicente
> *Sent:* Wednesday, December 15, 2021 11:27 AM
> *To:* OpenSIPS users mailling list
> *Subject:* [OpenSIPS-Users] how to change mysql data directory to CP
>
>
>
> Hello Everyone,
>
>
>
> I have changed my mysql directory and a had limited space on the default
> partition.   I applied the change and opensips is working fine.  But the
> control panel is not working and i can not find where to update it.
>
>
>
> Getting the following error:
>
>
>
>
>
> http://x.x.x.x/cp/login.php
>
> Error!: SQLSTATE[HY000] [2002] Can't connect to local MySQL server through
> socket '/var/lib/mysql/mysql.sock' (2)
>
>
>
> Any idea, where to update this?
>
>
>
> Saludos!
>
>
>
>
>
> Mario San Vicente
>
>
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Re: [OpenSIPS-Users] double sdp.

[Johan De Clercq]

Jeff, I think that I should call only once rtpengine_offer (then the sdp is
allright).
But I forgot to mention that the port that the A side sees should not
change meaning that whatever I respond on the received 183 should remain
the same on the 200 OK.
Therefore I wonder, if the fix wouldn't be to call only rtpengine_answer a
second time upon receiving the 200 OK from C.

Anyway, I will test that tomorrow morning and come back on this.

wkr,

Outlook voor iOS <https://aka.ms/o0ukef> downloaden
--
*Van:* Users  namens Jeff Pyle <
j...@ugnd.org>
*Verzonden:* Monday, October 18, 2021 11:50:22 PM
*Aan:* OpenSIPS users mailling list 
*Onderwerp:* Re: [OpenSIPS-Users] double sdp.

Johan,

To avoid problems like this, I call rtpengine_offer() in branch_route on
initial invites, and make sure to call rtpengine_delete() in any failure
route to remove any session from a failed offer that was never used.
Perhaps these will help in your situation as well.


- Jeff


On Mon, Oct 18, 2021 at 11:37 AM Johan De Clercq  wrote:

Hi,

A and B are on the same proxy.

A calls B,
(as I need transcoding I need to call rtpengine_offer here)
B returns 183 with SDP.
 (this implies calling rtpengine_answer in onreply_route)
 B lets the call time out
On the proxy I intercept the 480 returned by B
and I change the INVITE so that it point to SEMS
  (this ismplies calling rtpengine_offer again)

Issue: when you call 2x rtpengine_offer, you end up with a double sdp body.

So,
how can I instruct opensips to overwrite the body instead of appending one
?

wkr,
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[OpenSIPS-Users] double sdp.

[Johan De Clercq]

Hi,

A and B are on the same proxy.

A calls B,
(as I need transcoding I need to call rtpengine_offer here)
B returns 183 with SDP.
 (this implies calling rtpengine_answer in onreply_route)
 B lets the call time out
On the proxy I intercept the 480 returned by B
and I change the INVITE so that it point to SEMS
  (this ismplies calling rtpengine_offer again)

Issue: when you call 2x rtpengine_offer, you end up with a double sdp body.

So,
how can I instruct opensips to overwrite the body instead of appending one
?

wkr,
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Re: [OpenSIPS-Users] add SDP to 200ok without SDP

[Johan De Clercq]

No, in this scenario you don't need to repeat the sdp. 183 sdp=>200 no sdp.
And indeed update is then used to change it.  Ribbon works this way.

Wkr

On Wed, Sep 15, 2021, 20:21 Calvin Ellison  wrote:

> This doesn't answer your question: I'm curious if that is a broken
> behavior from the party sending the 200 without SDP. It is my understanding
> that all subsequent 18x and the 200 must use the same SDP if no UPDATE has
> taken place in the meantime. Please, correct me if that is a
> misunderstanding.
>
>
>
> Calvin Ellison
>
> Systems Architect
>
> calvin.elli...@voxox.com
>
> +1 (213) 285-0555
>
> 
>
> 
> 
> 
> 
>
> The information contained herein is confidential and privileged
> information or work product intended only for the individual or entity to
> whom it is addressed. Any unauthorized use, distribution, or copying of
> this communication is strictly prohibited. If you have received this
> communication in error, please notify me immediately.
>
>
> On Wed, Sep 15, 2021 at 10:02 AM Mario San Vicente 
> wrote:
>
>> Hello Everyone,
>>
>> has anyone face the following issue:  customer not accepting 200ok with
>> no SDP as valid.  Previously an 18[03] with SDP was sent, but the final
>> answer is 200 ok with no SDP from the termination leg.  So i was wondering
>> if there is a way to store the SDP and the append it to answering 200ok.
>>
>> Client   -->   Opensips as Proxy + rtpproxy  --> Termination Leg
>>
>>
>> Thank you in advance.
>> Mario SV
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Re: [OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

[Johan De Clercq]

No worries

On Wed, Sep 15, 2021, 11:43 Mark Allen  wrote:

> Ah! Thanks Răzvan and Johan. I was thinking that the options described in
> the OpenSIPS RTPEngine module documentation were the only ones
> available, but I missed the key phrase...
>
> *When passing an option that OpenSIPS is not aware of, it will be
> blindly sent to the rtpengine daemon to be processed.*
>
> ...I should be fine from here I think. Thanks for your help.
>
>
> On Tue, 14 Sept 2021 at 14:29, Răzvan Crainea  wrote:
>
>> I think the proper way to do it is
>>
>> `transcode-opus/48000/2/32000//maxplaybackrate--48000;stereo--0;useinbandfec--1;maxaveragebitrate--32000`
>> Anyway, as Johan suggests, follow the rtpengine documentation, you will
>> find more info there.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>> On 9/14/21 15:51, Mark Allen wrote:
>> > Hi Răzvan
>> >
>> > OpenSIPS passes the commands to RTPengine as you supply them, so in
>> > theory this should be supported. Did you try to set this up? Can you
>> > provide any logs about this?
>> >
>> >
>> > No. I'm not sure how I would specify this in the opensips.cfg
>> > rtpengine_offer parameters. From the documentation for the
>> > rptengine OpenSIPS module, it doesn't seem as if something like the
>> > following is an option...
>> >
>> >
>> >
>> transcode-opus='opus/48000/2/32000//maxplaybackrate=48000;stereo=0;useinbandfec=1;maxaveragebitrate=32000'
>>
>> >
>> >
>> > ...however, if I have tried this in the code (with or without quotes)
>> it
>> > doesn't produce an error, either from OpenSIPS or rtpengine, but
>> neither
>> > does it do anything - Opus fmtp in the SDP is not changed. I guess I'm
>> > looking for the correct OpenSIPS syntax to pass these parameters
>> > successfully to rtpengine
>> >
>> >
>> >
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Re: [OpenSIPS-Users] RTPEngine - changing codec format parameters from default values when transcoding - OpenSIPS 3.1

[Johan De Clercq]

Follow the rtpengine documentation. The necessary flags are given there,
just pass them in an avp to rtpengine_offer or answer

On Tue, Sep 14, 2021, 14:55 Mark Allen  wrote:

> Hi Răzvan
>
> OpenSIPS passes the commands to RTPengine as you supply them, so in
>> theory this should be supported. Did you try to set this up? Can you
>> provide any logs about this?
>>
>>
> No. I'm not sure how I would specify this in the opensips.cfg
> rtpengine_offer parameters. From the documentation for the
> rptengine OpenSIPS module, it doesn't seem as if something like the
> following is an option...
>
>
> transcode-opus='opus/48000/2/32000//maxplaybackrate=48000;stereo=0;useinbandfec=1;maxaveragebitrate=32000
> '
>
> ...however, if I have tried this in the code (with or without quotes) it
> doesn't produce an error, either from OpenSIPS or rtpengine, but neither
> does it do anything - Opus fmtp in the SDP is not changed. I guess I'm
> looking for the correct OpenSIPS syntax to pass these parameters
> successfully to rtpengine
>
>
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Re: [OpenSIPS-Users] Push SDP into Request

[Johan De Clercq]

wouldn't it be better to do first your manip and then pass it to rtpengine
?

Op wo 11 aug. 2021 om 12:39 schreef Mark Farmer :

> Hi everyone
>
> I am using rtpengine_offer() and storing the SDP in a variable which I
> then modify using the reg_exp transformation to delete a line.
>
> rtpengine_offer($var(PSTN_TPTY),,$var(SDPbody));
> $var(SDPbody) = $(var(SDPbody){re.subst,/a=silenceSupp:off\ -\ -\ -\
> -\r\n//g});
>
> How do I then push the resulting variable into the body of the INVITE?
>
> Many thanks
> Mark.
>
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Re: [OpenSIPS-Users] [FEATURE] Trace packets to file or syslog using OpenSIPS 3.2

[Johan De Clercq]

I agree fully with Giovanni

On Wed, Jul 7, 2021, 12:02 Giovanni Maruzzelli  wrote:

> On Wed, Jul 7, 2021 at 11:31 AM Răzvan Crainea 
> wrote:
>
>> Hi, All!
>>
>> Starting with OpenSIPS 3.2 you can have all the information captured by
>> the tracer module written to a local file. Moreover, you can also log
>> all the traced information to a syslog facility, and from there, store
>> it to a file, or even send it remotely using the syslog parameters.
>> All these thanks two the two new uri types added in OpenSIPS 3.2 to the
>> trace_id module: file and syslog[1].
>>
>>
> Wow, nice!!
>
> I'd like to be able to save to pcap too, it's my wet dream... (ok, ok, I
> go get a life :) )
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
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Re: [OpenSIPS-Users] [BLOG] Getting Started With Diameter In OpenSIPS 3.2

[Johan De Clercq]

Finally :-) bye bye radius.

Op do 17 jun. 2021 om 11:08 schreef Liviu Chircu :

> Hi folks,
>
> Our latest blog post [1] (10-minute read) features a thorough
> description of the Diameter protocol, the freeDiameter open-source
> project and how we integrated all of these into OpenSIPS 3.2 to offer
> accounting, authentication and future IMS capabilities.
>
> The blog also links to a detailed tutorial [2] on how you can set it up
> on your own and give it a test!
>
> [1]:
>
> https://blog.opensips.org/2021/06/17/getting-started-with-diameter-in-opensips-3-2/
> [2]: https://www.opensips.org/Documentation/Tutorials-Diameter-AAA
>
> Enjoy,
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
> OpenSIPS Summit 2021 Distributed | www.opensips.org/events
>
>
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Re: [OpenSIPS-Users] Getting header from 302

[Johan De Clercq]

that's because 302 is not an error.
So I guess that drop() is the only way.

Op wo 2 jun. 2021 om 15:42 schreef David Villasmil <
david.villasmil.w...@gmail.com>:

> Thanks Ben,
>
> That’s a good point. But only way I’ve found to jump over from oneply to
> failure_route is by doing a drop(). If there’s another way, I’d love to
> know about it!
>
> David
>
> On Wed, 2 Jun 2021 at 08:29, Ben Newlin  wrote:
>
>> You still don’t need to call drop() as long as you are handling the
>> request in failure_route. The 302 will not be sent back upstream as long as
>> failure_route either creates a new branch request or sends back a different
>> reply code. Only if failure_route exits without doing either of these
>> things would the downstream 302 be sent back upstream as-is.
>>
>>
>>
>> In fact, as far as I know drop() has no functionality for responses >=
>> 200.
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users  on behalf of Jeff Pyle <
>> j...@ugnd.org>
>> *Date: *Tuesday, June 1, 2021 at 2:48 PM
>> *To: *OpenSIPS users mailling list 
>> *Subject: *Re: [OpenSIPS-Users] Getting header from 302
>>
>> Oh!  Understood.
>>
>>
>>
>>
>>
>> - Jeff
>>
>>
>>
>>
>>
>> On Tue, Jun 1, 2021 at 2:42 PM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>> The thing is I _want_ to drop the 302, I don't want to do anything else
>> with it.
>>
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>> phone: +34669448337
>>
>>
>>
>>
>>
>> On Tue, Jun 1, 2021 at 6:46 PM Jeff Pyle  wrote:
>>
>> In my experience you don't need drop() in the reply route.  Just store
>> the AVP and move on.  Something like this:
>>
>>
>>
>> onreply_route[collect_identity] {
>>
>> if (is_present_hf("Identity")) {
>>
>> $avp(identity) := $hdr(Identity);
>>
>> setflag("GOT_IDENTITY");
>>
>> }
>>
>> }
>>
>>
>>
>> If you've armed both the reply and failure routes with t_on_reply() and
>> t_on_failure(), the $avp(identity) variable set here will be available in
>> the failure_route.  The GOT_IDENTITY flag, too.
>>
>>
>>
>>
>>
>> - Jeff
>>
>>
>>
>>
>>
>>
>>
>> - Jeff
>>
>>
>>
>>
>>
>> On Tue, Jun 1, 2021 at 11:20 AM David Villasmil <
>> david.villasmil.w...@gmail.com> wrote:
>>
>> Yes, I see it is documented.
>>
>>
>>
>> So the reply header is only availanble on the "onreply" route, not on the
>> "failure" route. That was my problem. I do indeed use an avp to store the
>> header.
>>
>> I ended up getting the header on the "onreply" and storing it in an avp,
>> set a flag and then drop(). I noticed the "failure" route is then executed.
>>
>> From there I can send the processing to the invite route and by  checking
>> the flag, adding the header from the avp.
>>
>>
>>
>> Thanks for your help!
>>
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>> phone: +34669448337
>>
>>
>>
>>
>>
>> On Tue, Jun 1, 2021 at 3:52 PM Ben Newlin  wrote:
>>
>> It’s documented that it works this way. The message being processed in
>> failure_route is the original request; in reply_route it’s the reply. [1]
>>
>>
>>
>> You can use variable context to access the reply from failure_route [2].
>> Another option would be to extract the header value into and AVP in
>> reply_route and then reference the AVP from failure_route.
>>
>>
>>
>>
>>
>> [1] - https://www.opensips.org/Documentation/Script-Routes-3-2
>>
>> [2] - https://www.opensips.org/Documentation/Script-CoreVar-3-2
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users  on behalf of David
>> Villasmil 
>> *Date: *Tuesday, June 1, 2021 at 10:43 AM
>> *To: *OpenSIPS users mailling list 
>> *Subject: *Re: [OpenSIPS-Users] Getting header from 302
>>
>> Yeah, my thing is when i use the failure route i can in theory grab the
>> response header and ignore the 302 and send to the invite route again to
>> actually send the call out via do_routing.
>>
>> What I'm trying to do is:
>>
>> - On receiving an invite: forward to an endpoint.
>>
>> - This endpoint will simply reply with 302 including a header.
>>
>> - I want to grab that header and continue routing normally (do_routing)
>>
>>
>>
>> I could do that with the failure route, but not so sure about the onreply
>> route.
>>
>>
>> Regards,
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.w...@gmail.com
>>
>> phone: +34669448337
>>
>>
>>
>>
>>
>> On Tue, Jun 1, 2021 at 2:34 PM Jeff Pyle  wrote:
>>
>> I don't think you're doing anything wrong.  I think I found the same
>> thing, that headers on the reply were available only in a reply route and
>> not in a failure route.  If you know where to look for them to populate the
>> AVP, I suppose it doesn't matter much.
>>
>>
>>
>> I haven't looked at the code but I suspect all the routes other than an
>> onreply_route give you access to the requests headers, and onreply_route
>> gives you access to the reply headers.  Makes sense I guess.
>>
>>
>>
>>
>>
>> - Jeff
>>
>>
>>
>>
>>
>> On Tue, Jun 1, 2021 at 9:31 AM David Villasmil <
>> 

Re: [OpenSIPS-Users] MS teams, reinvite after ACK

[Johan De Clercq]

remove Refer from your supported methods.
Do note that attended transfer will not work in this case.

wkr,

Op wo 2 jun. 2021 om 10:15 schreef Miha via Users :

> Hello
>
> i manage to fix this. I did not do t_relay() also seq Invites, after this
> everything works ok.
>
> Just on question, regarding transfers, i see that MS Teams send REFER in
> which trafter is defined. How do you deal with this? You do not allow REFER
> from MS teams and hope that MS teams will send new INVITE?
>
>
> thank you
> miha
>
> Jeff Pyle je 6/1/2021 ob 3:26 PM napisal:
>
> Miha,
>
> First, do you need to use "mtsbc.test.com:5060" in the first
> record_route_preset() param?  Can you use the IP address of your proxy
> instead?  FQDNs are legal of course, but outside of MS Teams'
> implementation, they're rarely required.  It's just another thing to go
> wrong.  Especially while testing.
>
> The ACK to the 200 OK is a sequential (in-dialog) request.  It's not part
> of the original INVITE transaction.  Your script will have a section like
>
> if (has_totag()) {
> if (loose_route()) {
> t_relay();
> }
> }
>
> for sequential requests through a loose-routing proxy.  This is very
> oversimplified and yours will have more.  In this section, however, is
> where you'll process the ACK because it has a to-tag (line 293) and a route
> header (line 298) so the conditions match.
>
> Use xlogs or the debug tool of your choice to diagnose what's happening in
> this section with the ACK.  In my scripts, I use global flag 0 to indicate
> when I want logging.  So, I might have something like this:
>
>if (has_totag()) {
>if (is_gflag(0)) xlog("L_NOTICE", " ...in-dialog $rm
> request\n");
># ...do all the things...maybe more logging like the line
> above...
>
>
> - Jeff
>
>
> On Tue, Jun 1, 2021 at 4:57 AM Miha via Users 
> wrote:
>
>> Hello
>>
>>
>> I have an issue and I am unable to find out what is wrong. Incoming calls
>> are working but when doing outbound call after 200OK, which is send to
>> Teams I get back ACK and after that Teams do again initial. I guess this is
>> not ok.
>>
>> I am doing this for outband calls:
>>
>>
>> xlog("L_INFO", "rtp rtps record route");
>> record_route_preset("mtsbc.test.com:5060","mtsbc.test.com
>> :5061;transport=tls");
>> add_rr_param(";r2=on");
>>
>> I am pasting here trace. Opensips is in the middle.
>>
>> Thank you for help!
>>
>> https://pastebin.com/qM0dMiCc
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Re: [OpenSIPS-Users] Generate CANCEL on 180

[Johan De Clercq]

Certain providers do this all the time : they bill faulty numbers

So they give 180 followed by instant 200 and then they play : "the number
that you tried to reach ".

Actually, Antonis, you should try a few of those numbers manually, I bet
you that they are all faulty.

Op di 20 apr. 2021 om 23:02 schreef Calvin Ellison :

> Once you receive 200 it is too late to CANCEL. You will need to ACK the
> 200 and then BYE the call.
>
>
> Regards,
>
> *Calvin Ellison*
> Systems Architect
> calvin.elli...@voxox.com
> +1 (213) 285-0555
>
> ---
> *voxox.com  *
> 5825 Oberlin Drive, Suite 5
> San Diego, CA 92121
> [image: Voxox]
>
>
> On Tue, Apr 20, 2021 at 1:14 PM Antonis Psaras 
> wrote:
>
>> I did the following
>>
>> if (t_check_status("180"))
>> {
>> t_cancel_branch();
>>drop;
>> }
>>
>> But there is an issue.
>>
>> When 180 is followed by 200 instantly, the CANCEL is not working as
>> expected.
>>
>> When I add a delay on Answer ie 1sec then CANCEL works.
>>
>> Any suggestion?
>>
>> Antonis Psaras
>>
>> -Original Message-
>> From: Users  On Behalf Of Kingsley Tart
>> Sent: Τρίτη, 20 Απριλίου 2021 20:10
>> To: OpenSIPS users mailling list 
>> Subject: Re: [OpenSIPS-Users] Generate CANCEL on 180
>>
>> Firstly, I'm new to OpenSIPS so treat my comments accordingly.
>>
>> But, can you do something in an onreply route?
>>
>> eg, in a test setup I have, when I get an INVITE I do this:
>>
>> create_dialog("pPB");
>> t_on_reply("doodle");
>>
>> (I can't remember whether the dialog is needed for this)
>>
>> and then I have this:
>>
>> onreply_route[doodle] {
>> # expect $T_reply_code to likely first be 100
>> # then 180 or 183 for a progressing call
>> # 200 when call is answered
>> # or failure code (eg 4xx) or whatever
>> if (t_check_status("^1[0-9][0-9]$")) {
>> switch ($T_reply_code) {
>> case 180: $acc_extra(t_ringing) = $Ts; break;
>> case 183: $acc_extra(t_progress) = $Ts; break;
>> }
>> } else if (t_check_status("^2[0-9][0-9]$")) {
>> $acc_extra(t_answer) = $Ts;
>> } else {
>> xlog("Something else\n");
>> }
>> }
>>
>> so when a 180 is received, it calls the above route function. Could you
>> send a CANCEL from there?
>>
>> Cheers,
>> Kingsley.
>>
>> On Tue, 2021-04-20 at 16:55 +0300, Antonis Psaras wrote:
>> > Dear all
>> >
>> > I am trying to create a service which will generate missed calls. In
>> > order to be more accurate, I want to CANCEL the request when 180 is
>> > received.
>> >
>> > The scenario is the following
>> >
>> > Asterisk Invite -> OpenSIPs -> Carrier
>> >
>> > Carrier 183 -> OpenSIPs -> Asterisk
>> >
>> > Carrier 180 -> OpenSIPs
>> >
>> > OpenSIPs Cancel -> Carrier
>> >
>> >
>> > Is that possible to be done from script without external app?
>> >
>> > Regards
>> >
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>>
>>
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Re: [OpenSIPS-Users] Handling mobile switch from wifi to data

[Johan De Clercq]

sending push before you send the invite solves many many problems.

Op wo 17 mrt. 2021 om 08:54 schreef Mark Allen :

> OK - thanks for that. I'll give it a try
>
> On Tue, 16 Mar 2021 at 12:22, Johan De Clercq  wrote:
>
>> Implement push.
>>
>> Op di 16 mrt. 2021 om 13:15 schreef Mark Allen :
>>
>>> We are using OpenSIPS 3.1 as a mid_registrar in AOR Throttling mode.
>>> When a mobile moves from data to wifi or vice versa, my understanding is
>>> that the registration on OpenSIPS becomes stale (because the phone has a
>>> new IP address). Does this mean that the mobile phone is not registered
>>> until registration expiry is reached on the mobile app and a new REGISTER
>>> is sent? If so, is there anything that can be done to mitigate this?
>>> ___
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Re: [OpenSIPS-Users] Handling mobile switch from wifi to data

[Johan De Clercq]

Implement push.

Op di 16 mrt. 2021 om 13:15 schreef Mark Allen :

> We are using OpenSIPS 3.1 as a mid_registrar in AOR Throttling mode. When
> a mobile moves from data to wifi or vice versa, my understanding is that
> the registration on OpenSIPS becomes stale (because the phone has a new IP
> address). Does this mean that the mobile phone is not registered until
> registration expiry is reached on the mobile app and a new REGISTER is
> sent? If so, is there anything that can be done to mitigate this?
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Re: [OpenSIPS-Users] Handling missing BYEs

[Johan De Clercq]

@Callum Guy  indeed :-)

Op wo 10 mrt. 2021 om 12:47 schreef Callum Guy :

> Might be a case of adding in a record_route() call in the appropriate
> place, hard to say without a trace :)
>
> On Wed, 10 Mar 2021 at 10:42, Mark Allen  wrote:
>
>> Hi Callum - thanks for that!
>>
>> Yes - it's generating the BYE at the Linux end but not sending it to the
>> remote OpenSIPS IP address but rather to an address on the local LAN -
>> hence the problem. Thanks for your help.
>>
>> Cheers,
>>
>> Mark
>>
>>
>>
>> On Wed, 10 Mar 2021 at 09:26, Callum Guy  wrote:
>>
>>> Hi Mark,
>>>
>>> It sounds like you may be having issues with the proxy not keeping
>>> itself in path for certain call scenarios.
>>>
>>> Are you able to provide a SIP trace and/or opensips config? Also if
>>> you're running Blink on a Linux system, can you get a SIP trace there to
>>> see if the BYE is being generated and sent somewhere else?
>>>
>>> Callum
>>>
>>> On Tue, 9 Mar 2021 at 16:32, Mark Allen  wrote:
>>>
 I'm seeing some odd behaviour which also leads into a broader question

 I have a NATed Blink app running on Linux on my home LAN. It connects
 to an OpenSIPS 3.1 server in on our office LAN which is a mid-registrar for
 an Asterisk server. I'm running sngrep on the OpenSIPS box to watch the
 traffic.

 If I call from the Blink app to another extension it all connects and
 audio works correctly. If I hangup in Blink, a BYE is sent via OpenSIPS to
 Asterisk - all good so far.
 If I call from another extension to the Blink app it all connects and
 audio works correctly. However, if I hangup in the Blink app, no BYE is
 sent to OpenSIPS.

 In most situations, this is merely inconvenient because, with the loss
 of RTP traffic, Asterisk generates a BYE after about 35 seconds to tidy
 everything up. However, if I'm doing an attended transfer, the BYE is
 needed to exit the call so that the transfer completes successfully. At the
 moment, if I hangup in the Blink app, there's a wait of 35 seconds until
 Asterisk creates the BYE before the call transfer is completed.

 While I'm mostly using Blink, I've seen similar failures to send BYEs
 from other apps. Does OpenSIPS offer anything that could help with this?
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Re: [OpenSIPS-Users] Retrieving parameters from file

[Johan De Clercq]

Gerwin, I will have a look.

Outlook voor iOS<https://aka.ms/o0ukef> downloaden

Van: Users  namens Gerwin van de Steeg 

Verzonden: Wednesday, February 10, 2021 5:26:10 AM
Aan: OpenSIPS users mailling list 
Onderwerp: Re: [OpenSIPS-Users] Retrieving parameters from file

Or you could use a sane and legible templating language.  I like the look of 
consul-template, but there are many others out there as the article suggests.. 
like Jinja2 or ERB.

On Wed, 10 Feb 2021 at 00:00, Johan De Clercq 
mailto:jo...@democon.be>> wrote:
Interesting, speaking for myself, it is absolutely team that I add m4 support.


Op di 9 feb. 2021 om 11:35 schreef Liviu Chircu 
mailto:li...@opensips.org>>:
On 09.02.2021 05:53, Dinesh Krishnamurthy via Users wrote:
> For example i would need to keep the IP Address/Port of the API
> Gateway which i communicate via rest_get without hardcoding as they
> would change based on the environment i am working with i.e.. staging,
> dev or production
>
> Please advise the way to do this.

Hi,

Sounds to me like your opensips.cfg file needs templating. If you are
using a 3.0 or newer OpenSIPS, this process is even further simplified,
as OpenSIPS natively integrates with any preprocessor that's out there.
See this guide [1] for more details.

Regards,

[1]: https://www.opensips.org/Documentation/Templating-Config-Files-3-2

--
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www.twitter.com/liviuchircu<http://www.twitter.com/liviuchircu> | 
www.opensips-solutions.com<http://www.opensips-solutions.com>


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Re: [OpenSIPS-Users] Retrieving parameters from file

[Johan De Clercq]

Interesting, speaking for myself, it is absolutely team that I add m4
support.


Op di 9 feb. 2021 om 11:35 schreef Liviu Chircu :

> On 09.02.2021 05:53, Dinesh Krishnamurthy via Users wrote:
> > For example i would need to keep the IP Address/Port of the API
> > Gateway which i communicate via rest_get without hardcoding as they
> > would change based on the environment i am working with i.e.. staging,
> > dev or production
> >
> > Please advise the way to do this.
>
> Hi,
>
> Sounds to me like your opensips.cfg file needs templating. If you are
> using a 3.0 or newer OpenSIPS, this process is even further simplified,
> as OpenSIPS natively integrates with any preprocessor that's out there.
> See this guide [1] for more details.
>
> Regards,
>
> [1]: https://www.opensips.org/Documentation/Templating-Config-Files-3-2
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
>
>
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Re: [OpenSIPS-Users] opensips dialplan regex

[Johan De Clercq]

if your expression means : any number of digits, then try something like
below:

try ([0-9]*)([A-Z]*)([0-9]*)

replace with \1\3.

Op di 2 feb. 2021 om 17:45 schreef volga629 via Users <
users@lists.opensips.org>:

> Hello Everyone,
> Here are some test result
>
> Feb  3 00:38:46 voice-proxy /usr/sbin/opensips[1337808]: [REQ_ROUTE]
> [INVITE] got incorrect 1506855JMGTJ4566 adjusting...
> Feb  3 00:38:46 voice-proxy /usr/sbin/opensips[1337808]: [REQ_ROUTE]
> [INVITE] new username -> [1506855JMGTJ4566] ~> 1506855
>
> As you see digits on the end got lost.
>
> volga629.
>
> On 2/2/21 11:47 AM, volga629 via Users wrote:
>
> Hello Everyone,
> I am trying clean up with dialplan  any characters from $rU except digits.
> I tried regex rule
>
> Matching Regular Expression
>  ([0-9]*)
>
> Substitution Regular Expression
> ([0-9]*)
>
> Replacement Expression
> \1
>
>
> Example:
> fhgg592199477719hh#
>
> I see that rule match regex, but rewrite is not happening.
>
> Any help thank you.
>
> volga629.
>
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Re: [OpenSIPS-Users] To-tag value in ACK

[Johan De Clercq]

is contact an fqdn ?
If not, look no further.

Op di 2 feb. 2021 om 15:06 schreef John Quick :

> I am seeing a problem in calls made to MS Teams via OpenSIPS configured as
> an SBC.
> The usual INVITE, 180, 183, 200 OK sequence looks okay, but the ACK request
> is not accepted by MS Teams.
> When I say "not accepted", I mean an ACK is sent to the Teams Proxy, but
> Teams responds after 30 seconds with BYE, reason is:
> "Call Controller timed out while waiting for acknowledgement"
>
> I am able to compare the sip trace for the failing call with a very similar
> call scenario on another Teams system that works.
> The only explanation I can suggest is that the UAC did not use the same
> value for the To-tag as it received in the 200 OK.
> Could this explain why Teams is failing to match it against the previous
> dialogue?
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
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Re: [OpenSIPS-Users] Dialogs with fix_nated_contact() have wrong RURI domain on sequential requests

[Johan De Clercq]

did you change the loose route part to fix route dialog ?

Op di 26 jan. 2021 om 16:39 schreef Jeff Pyle :

> Hello,
>
> This is on OpenSIPS nightly 3.1.1~20210125~8bab0da7b-1.
>
> I have a registrar configured with basic call routing between the
> registered AORs.  I use topology_hiding("D") to create the dialog on calls
> and normal stuff like has_totag() and topology_hiding_match() for
> sequential request handling.  All this seems fine.
>
> This appears high in the main route and appears to do exactly what it
> should:
>
> if (has_body("application/sdp")) {
> if (nat_uac_test(14)) {
> setflag("NAT_FLAG");
> }
> } else {
> if (nat_uac_test(6)) {
> setflag("NAT_FLAG");
> }
> }
>
> if (isflagset("NAT_FLAG")) {
> force_rport();
> if ($rm == "REGISTER") {
> fix_nated_register();
> } else {
> fix_nated_contact();
> }
> }
>
> And, for replies:
>
> onreply_route [handle_rtprelay_onreply] {
> # rtpengine and such, omitted for brevity
> if (isbflagset("NAT_BFLAG")) {
> fix_nated_contact();
> }
>
> exit;
> }
>
> When one client calls another, everything works fine.  lookup("location")
> works to update $rd with the original (private) Contact provided upon
> registration, and $du contains the actual received source IP:port to get to
> the device.  Excellent.  The INVITE goes out accordingly, and all is well.
>
> My problem occurs with sequential requests, say, re-INVITEs from on-hold
> events.  The dialogs themselves save the received IP:port values as the
> caller_contact and callee_contact values (from fix_nated_contact() above),
> so when the requests pass through the sequential handling section of the
> script and topology_hiding_match() does its fixups, the request URI domain
> of the relayed request has the received IP:port values of the target UA
> rather than the private IP:port values the UA provided during the initial
> request that established the dialog.
>
> I can't wrap my head around how to fix this.  The initial requests work
> because lookup() has the intelligence to distinguish the UAC's Contact from
> the received IP:port at REGISTER-time, but I can't see how to achieve this
> at dialog-creation time so sequential requests have the right RURI domain.
> Force the caller_contact and callee_contact to the private values somehow,
> and manage the route_set to point to the appropriate received IP:port?  I'm
> not sure how to configure that if it is the solution.
>
> Any direction would be appreciated!
>
>
> Regards,
> Jeff
>
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Re: [OpenSIPS-Users] How to see outgoing TLS OPTIONS

[Johan De Clercq]

Install Homer with portmirror on switch

On Thu, Jan 21, 2021, 13:31 Mark Farmer  wrote:

> Hi everyone
>
> In a recent thread I learnt that using trace() in local_route after
> appending a Contact header traces the message before the new header is
> actually added which means I don't get to see it.
>
> Is there a way to see the OPTIONS message after the new header is added?
>
> Many thanks
> Mark.
>
>
> On Thu, 12 Mar 2020 at 13:02, Mark Farmer  wrote:
>
>> I think I answered my own question :)
>>
>> Adding a sip_trace() into local_route seems to do the trick :)
>>
>> Mark.
>>
>>
>> On Thu, 12 Mar 2020 at 12:31, Mark Farmer  wrote:
>>
>>> Hi everyone
>>>
>>> I am using the drouting module to make SIP/TLS connections and I need to
>>> be able to capture the outgoing OPTIONS requests generated by drouting.
>>>
>>> I am thinking sip_trace("hep_dst", "d"); but where would I need to do
>>> that?
>>>
>>> Is there a better way?
>>>
>>> OpenSIPS 2.4
>>>
>>> Many thanks!
>>> Mark.
>>>
>>>
>>
>> --
>> Mark Farmer
>> farm...@gmail.com
>>
>
>
> --
> Mark Farmer
> farm...@gmail.com
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Re: [OpenSIPS-Users] Mediaproxy configuration

[Johan De Clercq]

I totally agree with the rtpengine suggestion

Outlook voor iOS downloaden

Van: Users  namens John Quick 

Verzonden: Thursday, January 21, 2021 10:40:18 AM
Aan: users@lists.opensips.org 
Onderwerp: Re: [OpenSIPS-Users] Mediaproxy configuration

Mark,

I recommend using rtpproxy (or possibly rtpengine) rather than mediaproxy
for your situation.
You need the address in the SDP to be the public IP 4x.xxx.xxx.xxx when it
is sending packets to the UAC but you need it to use its LAN address when
sending to the Asterisk server.
This is what bridge mode (or bridging mode) is used for, although the last
time I built a solution like this I didn't use bridge mode and instead
passed the relevant IP address as an argument when calling the rtpproxy
activation functions. Unfortunately, the latter approach means your
opensips.cfg script will need to be much more complicated.

I suspect your problem when using mediaproxy and advertised_ip =
4x.xxx.xxx.xxx is that it will pass that address to Asterisk in the SDP. In
which case, you might be able to get audio if you look at the network route
Asterisk would use to reach/connect to 4x.xxx.xxx.xxx and make sure the
mediaproxy relay is reachable. However, that does not sound like a good
solution to me - much better if Asterisk talks to the relay directly over
the LAN.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk



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Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

[Johan De Clercq]

Liviu,

For that case, why don’t you make an extra parameter ? E.g. 
use-broken-implementation 0|1

Outlook voor iOS downloaden

Van: Users  namens Liviu Chircu 

Verzonden: Tuesday, January 19, 2021 4:38:58 PM
Aan: OpenSIPS users mailling list 
Onderwerp: Re: [OpenSIPS-Users] Global variable $rm gives number when using 
$json

On 18.01.2021 20:53, Alex Kinch wrote:
I get this (identifying data redacted):

Jan 18 18:18:15 [33] request: INVITE from sip:XX@XX to sip:XX@XX
Jan 18 18:18:15 [33] Sending { "call_id": "942887463-1604195939-951239259", 
"ts": "2021-01-18 18:18:15Z", "src_ip": "XX", "dst_ip": "XX", "method": 1, 
"sip_from": "sip:XX@XX", "sip_to": "sip:XX@XX", "dialled": "XX" }

Any suggestions?


Hi Alex,

Thank you for the examples - indeed, that behavior is broken.

I just pushed a fix for this on "master" branch [1].  However, I'm a bit 
reluctant to backport it for the moment, because I haven't fully assessed its 
implications.  For example, could it be possible that people have already 
written code that relies on $json incorrectly returning the integer value of a 
variable which holds both a string and an integer, with string taking 
precedence (e.g. $rm)?

[1]: https://github.com/OpenSIPS/opensips/commit/6191f278a4

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Re: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on node failure

[Johan De Clercq]

Same here. Floating ip

On Tue, Jan 19, 2021, 16:23 Kevin Wormington 
wrote:

> That seems to be how most are setup…maybe I’m making it harder than it
> should be :-)
>
> Out of curiosity what did/do you use to monitor OpenSIPS as up for your
> failover or did you just rely on the IP (keepalived, etc.) reachability?
>
> Thanks,
>
> Kevin
> > On Jan 19, 2021, at 9:08 AM, Andy Dierlam  wrote:
> >
> > Ah, my setup was with a floating IP between servers.
> >
> > thanks
> > Andy
> >
> > On Tue, Jan 19, 2021 at 10:02 AM Kevin Wormington <
> kw...@missouri-telecom.com> wrote:
> > I’m not using a VIP and I have made some progress by setting a different
> active tag on each node…then upon node failure setting the failed node's
> tag to active on remaining node.  This lets the re-invite pinging work,
> etc.  It’s almost there but the handling of the BYE…they are still sent to
> the IP of the failed node even after re-invite pings so any in-progress
> calls from the failed node are zombie when they hang up until the re-invite
> ping times out (30 seconds).   I found an article about initiating a
> re-invite on the new node with something like "opensips-cli -x mi
> dlg_send_sequential callid="442CB6C1-6005F8B80009DA08-FC731700"
> mode=challenge body=outbound” but that either seems to terminate the call
> immediately or say the dialog wasn’t found.
> >
> >
> > Thanks,
> >
> > Kevin
> > > On Jan 19, 2021, at 8:46 AM, Andy Dierlam 
> wrote:
> > >
> > > With dialog writing to db that both servers use.   And same tag on
> both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
> > > had this working on opensips 2.4
> > >
> > > thanks
> > > Andy
> > >
> > >
> > > On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <
> kw...@missouri-telecom.com> wrote:
> > > Hi,
> > >
> > > I've been attempting to get a two node active/active setup to work
> with the v3.1 clusterer module sharing usrloc and dialog.  The setup is
> fronted by a proxy that handles all of the NAT/media so either OpenSIPS
> instance can communicate directly with the user.
> > >
> > > What I have working so far:
> > >
> > > Registrations and calls work when sent to either node and if you stop
> OpenSIPS on a node new calls work fine using the other node.
> > >
> > > What I can’t get to work:
> > >
> > > Calls that are already in progress to switch between nodes when one
> node fails.
> > >
> > >
> > > I have messed around with various sharing tags…no tag, same tag,
> different tags but haven’t had any luck.   I’m guessing that I’m missing
> something to trigger the remaining node to send re-invites.  Has anyone
> attempted this type of setup and have any ideas?
> > >
> > > Thanks,
> > >
> > > Kevin
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Re: [OpenSIPS-Users] PRACK response to 183 without SDP

[Johan De Clercq]

you mean that the remote party sends you required:100rel ? The only way
that I know is by using b2b logic.
Maybe Bogdan can shed some light here.

Op ma 18 jan. 2021 om 17:07 schreef solarmon :

> Hi,,
>
> I have a requirement to stop 183 without SDP packets from being passed, as
> well as having to reply back with a PRACK.
>
> I can stop the 183 without SDP from being passed on with the following in
> onreply_route[]
>
>  if ($rs == "183" && !has_body_part("application/sdp")) {
> drop();
> exit;
> }
>
> However, how do I reply back with a PRACK?
>
> Thank you.
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Re: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

[Johan De Clercq]

Firewall is not sip aware, rtprelay via box in dmz

Outlook voor iOS downloaden

Van: Users  namens Mark Allen 

Verzonden: Wednesday, January 13, 2021 5:08:27 PM
Aan: OpenSIPS users mailling list 
Onderwerp: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

Hi all - I've been banging my head against this but not succeeding.

Our setup...

UAC   192.168.x.x
  |
Router5.x.x.x
  |
(internet)
  |
Firewall  46.x.x.x maps
  |   directly to
OpenSIPS  192.168.x.x  Mid-registrar
  |
Asterisk  192.168.x.x


Current situation:
- UAC can register on Asterisk via OpenSIPS
- UAC can call destination registered on Asterisk on local n/w to Asterisk box
- Destination extension rings and can pick up call
- There is no audio either way & call drops after about 30 secs (Asterisk kills 
call with "Requested channel not available" because not RTP traffic is reaching 
destination)

I have tried passing audio through Mediaproxy on OpenSIPS box but with no 
success. Using Wireshark I can see RTP traffic initiated at both ends, but it 
doesn't reach the other end either way.

Is there some definitive guide to setting this up correctly or are there 
specific steps that I need to follow?

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Re: [OpenSIPS-Users] sangoma issue ?

[Johan De Clercq]

Why not use rtpengine to do transcoding?

Outlook voor iOS downloaden

Van: Users  namens Andy Dierlam 

Verzonden: Wednesday, January 13, 2021 4:26:03 PM
Aan: users@lists.opensips.org 
Onderwerp: Re: [OpenSIPS-Users] sangoma issue ?

forgot to mention this from:

opensips -V
version: opensips 3.0.4 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, 
F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: d19b20e
main.c compiled on 14:39:30 Dec 21 2020 with gcc 4.8.5

FYI get the same results from 2.4, 3.0, and 3.1





On Tue, Jan 12, 2021 at 3:14 PM Andy Dierlam 
mailto:adier...@ptgi-ics.com>> wrote:
Hello All,

Wondering if anyone can offer some guidance on an issue seemingly with a newly 
setup Sangoma D-500 Transcoding card.

Issue:
When using transcoding, soon after 1 transcoded call completes, utimer task 
messages until opensips is restarted
seems like it happens when fetch_dlg_value: looking for 


Jan 12 15:05:26 [21578] DBG:tm:timer_routine: timer routine:2,tl=0x7f9ce5fd3cc0 
next=(nil), timeout=16

Jan 12 15:05:26 [21578] DBG:tm:wait_handler: removing 0x7f9ce5fd3c40 from table

Jan 12 15:05:26 [21578] DBG:tm:delete_cell: delete transaction 0x7f9ce5fd3c40

Jan 12 15:05:26 [21578] DBG:tm:run_trans_callbacks: trans=0x7f9ce5fd3c40, 
callback type 4096, id 3 entered

Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: destroying dialog 0x7f9ce5fcdc50

Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: dlg expired or not in list - 
dlg 0x7f9ce5fcdc50 [57:1450509359] with clid 
'3b72b48a67a1eec840e2e1db12f8b37f@x.x.x.x' and tags 'as6d2cf831' 
'Yjg9IHv3tknz-KgDTUF.K-lleno6X-hI'

Jan 12 15:05:26 [21578] DBG:dialog:run_dlg_callbacks: dialog=0x7f9ce5fcdc50, 
type=2048

Jan 12 15:05:26 [21578] DBG:dialog:fetch_dlg_value: looking for 

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task  
already scheduled 100 ms ago (now 16950 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task  
already scheduled 200 ms ago (now 17050 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task  
already scheduled 300 ms ago (now 17150 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task  
already scheduled 400 ms ago (now 17250 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task  
already scheduled 490 ms ago (now 17340 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task  
already scheduled 590 ms ago (now 17440 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task  
already scheduled 690 ms ago (now 17540 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task  
already scheduled 790 ms ago (now 17640 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task  
already scheduled 890 ms ago (now 17740 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:timer_ticker: timer task  
already scheduled 990 ms ago (now 17740 ms), delayin



opensips-cli trap output ( not sure what am looking at here, in case helps )


#1  0x7f9cde9aee5f in sangoma_worker_loop (proc_no=) at 
sngtc_proc.c:46

req = {type = REQ_FREE_SESSION, response_fd = 78, sng_req = {usr_priv = 
0x7f9ce8c8f86b , tag = 15, rtcp_enable = 5 '\005', a = {codec_

id = 32668, ms = 20, host_ip = 0, host_netmask = 2021, host_udp_port = 0}, b = 
{codec_id = 9217376, ms = 0, host_ip = 3905485451, host_netmask = 32668, host_

udp_port = 20}}, sng_reply = 0x7f9ce5fd2c78}

rc = 0

__FUNCTION__ = "sangoma_worker_loop"

#2  0x005045c8 in start_module_procs () at sr_module.c:858

m = 0x7f9ce7beafa8

n = 0

l = 0

x = 

__FUNCTION__ = "start_module_procs"

#3  0x0041e3df in main_loop () at main.c:779

startup_done = 0x0

chd_rank = 0

last_check = 0

rc = 

#4  main (argc=, argv=) at main.c:1479

c = 

r = 0

tmp = 0x1 

tmp_len = 

port = 

proto = 

protos_no = 

options = 0x664908 "f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:p:P:G:W:o:a:k:s:"

ret = -1

seed = 4061548656

rfd = 

__FUNCTION__ = "main"

thanks
Andy
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Re: [OpenSIPS-Users] OpenSIPS as simple SIP proxy

[Johan De Clercq]

take the residential config and see where that gets you.

Op wo 13 jan. 2021 om 08:21 schreef Conor Power :

> Hi,
>Apologies for the noob question but I'm hoping someone can point me in
> the right direction.
>
> I am trying to use OpenSIPS as a simple proxy to proxy all calls to
> another SIP endpoint and back again to the original client. The only role
> of the OpenSIPS server is to function as the proxy and it is for all
> requests.
>
> I have OpenSIPS up and running and can see the requests coming inbound
> using ngrep but I've had no success proxying the requests.
>
> I added a sethostport() call in the config file but really am not sure
> where or how it fits in.
>
> If someone might point me to a simple config file that would be used for
> such a proxy setup, it would be greatly appreciated.
>
> Conor
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Re: [OpenSIPS-Users] call api

[Johan De Clercq]

Thanks Razvan, that's exactly the info that  I was looking for.
call api is very interesting to create an ACD like opensips instance.

Op ma 11 jan. 2021 om 11:08 schreef Răzvan Crainea :

> Hi, Johan!
>
> Check out this repo[1].
>
> [1]
>
> https://github.com/razvancrainea/opensips-summit-distributed-2020/blob/call-api/opensips.cfg
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 1/8/21 5:08 PM, johan wrote:
> > Hello Razvan,
> >
> >
> > is there already an opensips.cfg example config available for using call
> > api ?
> >
> >
> > wkr,
> >
> >
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Re: [OpenSIPS-Users] Transparent TLS

[Johan De Clercq]

Anycast can in my opinion only work in IP6.

Op ma 11 jan. 2021 om 09:35 schreef Giovanni Maruzzelli :

> Hi Yavari,
>
> On Sat, Jan 9, 2021 at 8:03 AM H Yavari via Users <
> users@lists.opensips.org> wrote:
>
>> Clients must have TLS connection and we have an OpenSIPS cluster on the
>> front of Asterisk servers. So in this case, if client's connection with one
>> SIP proxy node goes down, it should be re-establish with other node in
>> cluster? or as all cluster nodes are using shared DB and they talk to each
>> other via BIN, client connection remains? thanks.
>>
>>
> I do not think there is a way to have TCP (TLS, WebRTC, etc) connection to
> survive a server failover.
>
> You may want to have the clients to re-connect (reregister and reinvite)
> in case of failover.
>
> Or, maybe clustering OpenSIPSs in active-active via anycast.
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
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Re: [OpenSIPS-Users] Weird behaviour in B2BUA with provisional media

[Johan De Clercq]

Opensips is not a full b2b, use something else for refer

On Thu, Dec 17, 2020, 12:25 Virgílio Cunha  wrote:

> Hello everyone,
>
> I have a problem in B2B behaviour with provisional media configured.
>
> In my scenario, I have a call to a call center that transfers the call
> (REFER) to another one. So, as I have a provisional media configured, a new
> SIP leg is generated by B2B between the user terminal and the media server
> until the call center answers it.
>
> Up to this point the call doesn't show any problem, but if the user
> terminal does a reINVITE to negotiate the SDP, I see weird behaviour.
>
> In the provisional media SIP leg, everything seems right. The media server
> answers to the reINVITE with 200 OK, but the 200 OK never was generated by
> B2B on the terminal SIP leg, instead of it I saw a new ACK forwarded to the
> user terminal (should be a 200 OK) and the B2B generates a new SIP leg to
> same call center referred previously.
>
> Why was a new SIP leg generated to the referred destination? If there's
> not any reinvite from the user terminal the REFER scenario doesn't have any
> problem.
>
> Does anyone know if this behavior was fixed in the latest version?
> Is there any additional configuration that I can do to fix this behaviour?
>
> I am currently using opensips 3.0.2.
>
> Best Regards,
> Virgílio Cunha
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Re: [OpenSIPS-Users] too many parameters for command

[Johan De Clercq]

yeah indeed, or send them over in a ; separated list and then split them in
the python function.

wkr,

Op ma 7 dec. 2020 om 07:54 schreef Tomi Hakkarainen :

> Hi,
>
> I would pass params in a python dictionary if its alllowed.
>
> Tomi
>
> On 6. Dec 2020, at 14.35, Elaine Huang  wrote:
>
> 
> Hi Vlad,
>
> Thanks.
>
> What would you do if you need to pass multiple parameters to a function?
> I'm thinking of compile the params into one string (separated by comma
> maybe) and parse them in the python function, but that's hacky.
>
> On Sat., 5 Dec. 2020, 03:29 Vlad Patrascu,  wrote:
>
>> Hi Elaine,
>>
>> Unfortunately the documentation is actually misleading and the function
>> accepts only one extra argument for passing to python.
>>
>> Regards,
>>
>> --
>> Vlad Patrascu
>> OpenSIPS Developerhttp://www.opensips-solutions.com
>>
>> On 04.12.2020 11:30, Elaine Huang wrote:
>>
>> Hi,
>>
>> I wrote a python function that takes 3 parameters (excluding msg):
>>
>>
>> class MyClass:
>>   …
>>   my_f(self, msg, param1, param2, param3):
>> …
>>   …
>>
>>
>> While the python module readme suggests it can accept extra args (more
>> than 1), opensips fail to start with error: too many parameters for command
>> 
>>
>> my config code:
>> python_exec("my_f", param1, param2, param3)
>>
>>
>> It can start if I change it to:
>> python_exec("my_f", param1)
>>
>>
>> Any idea why?
>>
>>
>> OpenSIPS version: 3.1
>>
>>
>>
>> Kind Regards,
>> Elaine
>>
>>
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Re: [OpenSIPS-Users] lua_exec in timer route - OpenSIPS 3.1

[Johan De Clercq]

what you can try, is to call another route in the time route.
And then in that route, you execute the lua script.
maybe (just a myabe) that will work.

wkr,

Op do 3 dec. 2020 om 12:23 schreef Mark Allen :

> Hi Johan
>
> In the documentation for 3.1 lua module - TIMER_ROUTE is not one of the
> routes available to lua_exec. If I include it in a TIMER route, OpenSIPS
> fails to start with syntax error and the log error is:
>
> CRITICAL:core:yyerror: parse error in
> /etc/opensips/opensips.cfg:265:19-20: Command  cannot be used in
> the block#012
>
> If I move the lua_exec command into main route{ it works fine
>
> I also encounter the problem running a cache_remove_chunk in a TIMER route
> although the documentation doesn't say that it's not valid for TIMER route.
> It fails on startup with the error:
>
> CRITICAL:core:yyerror: parse error in
> /etc/opensips/opensips.cfg:266:33-34: Command  cannot
> be used in the block#012
>
> Again - if I run the command in main route{ the command works fine
>
>
> cheers,
>
> Mark
>
> On Thu, 3 Dec 2020 at 11:01, Johan De Clercq  wrote:
>
>> It for sure does not run in async mode.
>> Did you try executing a script in timer route ?
>> What's the output in the log ?
>>
>> Op do 3 dec. 2020 om 11:56 schreef Mark Allen :
>>
>>> Is there a way to run a lua_exec from a timer route?
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Re: [OpenSIPS-Users] lua_exec in timer route - OpenSIPS 3.1

[Johan De Clercq]

It for sure does not run in async mode.
Did you try executing a script in timer route ?
What's the output in the log ?

Op do 3 dec. 2020 om 11:56 schreef Mark Allen :

> Is there a way to run a lua_exec from a timer route?
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Re: [OpenSIPS-Users] Regex to test variable

[Johan De Clercq]

A classic mistake, also very often mate in C.

Op do 26 nov. 2020 om 11:04 schreef Mark Farmer :

> Thanks Donat, good tip :)
>
> Turned out to be a missing equals sign
>
> if $avp(dtmf_compat)= "0" {
>
> if $avp(dtmf_compat) == "0" {
>
> Silly me, took me ages to spot that!
>
> Thanks again for the tip!
>
> Mark.
>
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Re: [OpenSIPS-Users] Opensips-cp 8.3.0 HTTP/1.1 400 Bad Request..

[Johan De Clercq]

You need to use mi_json

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Van: Users  namens Mario San Vicente 

Verzonden: Wednesday, October 28, 2020 8:06:09 PM
Aan: OpenSIPS users mailling list 
Onderwerp: [OpenSIPS-Users] Opensips-cp 8.3.0 HTTP/1.1 400 Bad Request..

Hello Everyone,

Here i have a quick question.

I installed opensips-cp 8.3.0 in a Centos installation from sources version: 
opensips 3.1.

I seems to me that  the configuration is fine, mysql DB, and users.

But Opensips-cp can't speak MI to opensips.

I get the following at trace level:

T 127.0.0.1:34550 -> 
127.0.0.1: [AP] #144
  POST /json HTTP/1.1..Host: 127.0.0.1:..Accept: */*..Content-Type: 
application/json..Content-Length: 
78{"jsonrpc":"2.0","id":1,"method":"dlg_lis
  t","params":{"index":0,"counter":20}}w.._
#
T 127.0.0.1: -> 
127.0.0.1:34550 [A] #145
  8...w.._XZ..
#
T 127.0.0.1: -> 
127.0.0.1:34550 [AP] #146
  HTTP/1.1 400 Bad Request..Content-Length: 46..Content-Type: text/html..Date: 
Wed, 28 Oct 2020 18:59:03 GMTUnable to parse URL!w.._C...
#

And at php level iget:

 [:error] [pid 17689] [client x.x.x.x:55303] PHP Warning:  Creating default 
object from empty value in 
/var/www/html/opensips-cp/config/tools/system/dialog/local.inc.php on line 27, 
referer: http://x.x.x.x/cp/tools/system/dialog/dialog.php?action=refresh
 [:error] [pid 17689] [client x.x.x.x:55303] PHP Notice:  Use of undefined 
constant CURLINFO_RESPONSE_CODE - assumed 'CURLINFO_RESPONSE_CODE' in 
/var/www/html/opensips-cp/web/common/mi_comm.php on line 50, referer: 
http://x.x.x.x/cp/tools/system/dialog/dialog.php?action=refresh
[:error] [pid 17689] [client x.x.x.x:55303] PHP Warning:  curl_getinfo() 
expects parameter 2 to be long, string given in 
/var/www/html/opensips-cp/web/common/mi_comm.php on line 50, referer: 
http://x.x.x.x/cp/tools/system/dialog/dialog.php?action=refresh
[:error] [pid 17689] [client x.x.x.x:55303] PHP Warning:  array_key_exists() 
expects parameter 2 to be array, null given in 
/var/www/html/opensips-cp/web/common/mi_comm.php on line 61, referer: 
http://x.x.x.x/cp/tools/system/dialog/dialog.php?action=refresh

 php -v
PHP 7.3.24 (cli)


Anyone have any idea why it fails??

Thank you very much
--
Mario SV
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Re: [OpenSIPS-Users] rtpengine_delete not returning immediately

[Johan De Clercq]

Would it be an option to disable rtcp?

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Van: Users  namens John Quick 

Verzonden: Wednesday, October 28, 2020 2:34:25 PM
Aan: users@lists.opensips.org 
Onderwerp: Re: [OpenSIPS-Users] rtpengine_delete not returning immediately

A packet capture revealed some interesting facts about this problem.
There is a genuine delay in the response from rtpengine when the delete command 
is sent. In the packets I captured, the delay was 2 seconds.
It is possible this delay arises because I am using a low spec virtual server 
with very little memory to host rtpengine.
Alternatively, it could be that rtpengine is doing additional processing of the 
audio streams so it can send a report back to OpenSIPS.
Using wireshark to inspect the packets, I can see that the response contains 
round-trip times, packet loss, average/low/high MOS scores, etc.

Does anyone know if there is a way to:
 a) Reduce the delay within rtpengine so it responds quicker to the delete 
command
 b) To make calls to rtpengine_delete() act asynchronously so they don't block 
the main SIP processing child threads?

John Quick
Smartvox Limited



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Re: [OpenSIPS-Users] rtpengine_delete not returning immediately

[Johan De Clercq]

Did you open an issue on github on rtpengine? Rfuchs comments are always 
enlightening

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Van: Users  namens John Quick 

Verzonden: Wednesday, October 28, 2020 12:51:33 PM
Aan: users@lists.opensips.org 
Onderwerp: Re: [OpenSIPS-Users] rtpengine_delete not returning immediately

Callum,

Thanks for the suggestion. I added a rule to iptables on the OpenSIPS server to 
allow all UDP from the rtpengine server.
Unfortunately, it made no difference. I also checked the route using mtr and it 
looks fine - less than 5 ms latency, 1 drop in >4000.
Network connectivity seems unlikely to be a contributory factor because the 
offer and answer commands work exactly as I would expect.
Only the delete command exhibits a delay returning.

John Quick
Smartvox Limited


> From: Callum Guy 
> Sent: 27 October 2020 21:12
> To: John Q ; OpenSIPS users mailling list 
> 
> Subject: Re: [OpenSIPS-Users] rtpengine_delete not returning immediately
>
> Have you double checked it's not a firewall issue on the SIP proxy?
>
> The transport is typically UDP so there is a fair chance it's blocked.


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Re: [OpenSIPS-Users] Fwd: Retransmission and 300 Redirect

[Johan De Clercq]

Create dialog and then t check trans

On Thu, Oct 22, 2020, 11:35 Pavel Eremin  wrote:

>
> Hi, community!
> Is anyone can help me to understand how to avoid 2 situations:
>
> 1. I don't want to send retransmission  (INVITE) from "opensips" to
> "redirect" server, even "redirect" does not answer.
>
> 2. If I have sent a few invites to "redirect" , then i will receive a few
> 300 Redirect messages, and "opensips" will generate branches for all
> contacts in all "300 Redirect" messages. how can i ignore all "300
> Redirects".
>
> I am not sure that t_check_trans() can help me with that.
> Thanks. I added a picture with the call flow I have.
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Re: [OpenSIPS-Users] does lua_exec supports async and launch

[Johan De Clercq]

yes indeed, that will take a while to make this working.

Op do 15 okt. 2020 om 10:33 schreef Răzvan Crainea :

> No, not fdopen, that will just open a FILE* out of a file descriptor.
> The async mechanism works directly with a file descriptor.
> I've quickly checked the lua API, and I couldn't see any mechanisms that
> you could use - in this case, you should probably "simulate" an external
> process that gets "lua jobs" and executes them asynchronously, and
> returns a file descriptor that can be tracked down to see the whether
> the exec completed. I'll have to admit, this doesn't see easy :(.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 9/29/20 7:50 PM, johan wrote:
> > You refer to fdopen in Ansi C ?
> >
> > On 28/09/2020 09:43, Răzvan Crainea wrote:
> >> Hi, Johan!
> >>
> >> Unfortunately the lua module does not support asynchronous operations.
> >>
> >> TBH, I'm not that familiar with the module, so I really don't know
> >> exactly how it gets executed, but in order to add async support, we
> >> need the lua mechanism to provide us a file descriptor (or somehting
> >> similar) that get signaled when an operation is completed. Or at least
> >> provide some mechanisms to run scripts asynchronous and provide a
> >> callback that indicates when the script has finished.
> >>
> >> Best regards,
> >>
> >> Răzvan Crainea
> >> OpenSIPS Core Developer
> >> http://www.opensips-solutions.com
> >>
> >> On 9/25/20 11:34 AM, johan wrote:
> >>> does lua_exec supports async calls ?
> >>>
> >>> If no, can you please indicate what needs to be added to have this
> >>> support ?
> >>>
> >>>
> >>> wkr,
> >>>
> >>>
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Re: [OpenSIPS-Users] does lua_exec supports async and launch

[Johan De Clercq]

fine, then i have a start to add it.

On Mon, Sep 28, 2020, 09:47 Răzvan Crainea  wrote:

> Hi, Johan!
>
> Unfortunately the lua module does not support asynchronous operations.
>
> TBH, I'm not that familiar with the module, so I really don't know
> exactly how it gets executed, but in order to add async support, we need
> the lua mechanism to provide us a file descriptor (or somehting similar)
> that get signaled when an operation is completed. Or at least provide
> some mechanisms to run scripts asynchronous and provide a callback that
> indicates when the script has finished.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 9/25/20 11:34 AM, johan wrote:
> > does lua_exec supports async calls ?
> >
> > If no, can you please indicate what needs to be added to have this
> > support ?
> >
> >
> > wkr,
> >
> >
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Re: [OpenSIPS-Users] Setting sdp

[Johan De Clercq]

use rtpengine and read the documentatiin attentively.

On Wed, Sep 23, 2020, 15:36 Abisai Matangira  wrote:

> Hello ,
>
> I have an opensips instance with two interfaces
>
> One facing upstream switch
>  facing a group of my wholesale customers
>
> When I receive invite methods  from the second interface facing wholesale
> customers , it replies giving the sdp c address of the first interface
> instead of the second interface
>
>
>
> On another note may I be added on the slack channel if ok.
>
>
>
>
>
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Re: [OpenSIPS-Users] store data locally

[Johan De Clercq]

I tend to agree with Liviu.  Therefore Ovidiu, please convince us of the
advantages of db_text :-)

Op wo 23 sep. 2020 om 07:58 schreef Liviu Chircu :

> On 22.09.2020 18:42, Ovidiu Sas wrote:
> > If you don't want to run a full blown db, then you can use db_text
> > without cacheDB.
> > The data is cached into memory at startup. If you update the text
> > file, you can re-cache the data [1].
>
> Hi, Ovidiu!
>
> May I segue into discussing some questions that have been puzzling me
> for a while now?
>
>  "In 2020, why should developers choose DB TEXT over DB SQLITE?
> Shouldn't DB TEXT be obsoleted?"
>
> As far as history goes, it seems DB TEXT was created by Daniel in 2003.
> Which, in my opinion, even for that time, it seems like a "reinvent the
> wheel" kind of effort, since SQLite had already been GA'ed for almost 3
> years [1].
>
> In order to come up with an answer, we can break down my original
> questions:
>
> * DB TEXT and SQLite both aim to be lightweight, serverless, file-based
> SQL databases with RAM caching.  True or false?
> * does DB TEXT have any features that SQLite doesn't?
> * does DB TEXT support a richer SQL syntax than SQLite?
> * does DB TEXT have less bugs than SQLite?
> * does DB TEXT handle in-memory caching better than SQLite?
> * does DB TEXT handle disk files better than SQLite?
>
> Thank you in advance for the discussion!
>
> Best regards,
>
> [1]: https://sqlite.org/changes.html
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
>
>
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Re: [OpenSIPS-Users] TLS Error

[Johan De Clercq]

it seems to me that it can't load your certificate.

Op do 17 sep. 2020 om 15:16 schreef Andrew Colin :

> Hi Guys
>
> I am trying to get tls to work but getting some errors.
> i am using letsencrypt and opensips 3.1
>
> my config is
>
> loadmodule "proto_tls.so"
>
>
> loadmodule "tls_mgm.so"
>
>
> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom")
>
>
> modparam("tls_mgm", "server_domain", "dom1")
>
> modparam("tls_mgm", "match_ip_address", "[dom1]myip:5061")
>
> modparam("tls_mgm", "match_sip_domain", "[dom1]mydomain.co.uk")
>
>
>
> modparam("tls_mgm", "tls_method", "[dom1]TLSv1_2")
>
> modparam("tls_mgm", "verify_cert", "[dom1]1")
>
> modparam("tls_mgm", "require_cert", "[dom1]1")
>
> modparam("tls_mgm", "certificate", "[dom1]/etc/letsencrypt/live/
> mydomain.co.uk/cert.pem")
>
> modparam("tls_mgm", "private_key", "[dom1]/etc/letsencrypt/live/
> mydomain.co.uk/privkey.pem")
>
> modparam("tls_mgm", "ca_list", "[dom1]/etc/letsencrypt/live/
> mydomain.co.uk/cert.pem")
>
> modparam("tls_mgm", "ca_dir", "[dom1]/etc/letsencrypt/live/bmydomain.co.uk
> ")
>
>
>
> but i get this error
>
>
>
> INFO:tls_mgm:mod_init: disabling compression due ZLIB problems
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]: INFO:tls_mgm:init_tls_dom:
> Processing TLS domain 'dom1'
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]:
> NOTICE:tls_mgm:init_tls_dom: No EC curve defined
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]:
> INFO:tls_mgm:get_ssl_ctx_verify_mode: client verification activated. Client
> certificates are mandatory.
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]:
> NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]:
> ERROR:tls_mgm:load_certificate: unable to load certificate file
> '/etc/letsencrypt/live/mydomain.co.uk/cert.pem'
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]:
> ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'dom1'
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]: ERROR:core:init_mod:
> failed to initialize module tls_mgm
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]: ERROR:core:main: error
> while initializing modules
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]: INFO:core:cleanup: cleanup
>
> Sep 17 12:59:41 proxy /usr/sbin/opensips[8155]: NOTICE:core:main:
> Exiting
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Re: [OpenSIPS-Users] wait_for_event(event,filter,timeout)

[Johan De Clercq]

More or less the same problem here.
I need to trigger on receiving a register after a push notification, I do
see the REGISTER arriving but nothing happens.
I use 3.1.  Might it be that we have a module issue ?

Op do 20 aug. 2020 om 08:41 schreef Darpan Patel <
darpan.gabani1...@gmail.com>:

> Hello All , i have a query regarding wait_for_event functionality .In
> documentation usage of event is like this :
>
> # wait for callee to register
> $avp(filter) = "aor="+$rU+"@"+$rd
> async( wait_for_event("E_UL_AOR_INSERT","$avp(filter)", "40"),  resume_call);
> # done
> ...
> route[resume_call] {
>   xlog("user $avp(aor) is now registered\n");
>   lookup("location");
>   t_relay();
> }
>
> But in my case after 40 seconds it's not trigger resume_call route, so 
> resume_call only called after the event will succeed ? I want to implement a 
> feature like if callee is not registered till 40 seconds then relay call to 
> PSTN Gateway .thanks alot in advance .
>
> regards ,
>
> Darpan Patel
>
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Re: [OpenSIPS-Users] Intercept 477 on TCP closed

[Johan De Clercq]

use t_relay wih 0x2 option.

On Mon, Aug 17, 2020, 15:16 Callum Guy  wrote:

> Hi All,
>
> Using OpenSIPs 3.0.3
>
> I'm dealing with a client device with a faulty network, they are using a
> softphone WebRTC client and the TCP connections disappear sporadically.
>
> When the media server issues a RE-INVITE session timer OpenSIPs discovers
> the closed TCP connection and returns 477 to the media server. In this case
> the media server is FreeSWITCH which promptly ignores the
> non-standard session timer response and the call hangs. I want to close the
> call immediately in this situation as there is no way I can see to
> reestablish the connection. The call is typically bridged on the media
> server so the other call leg is left dangling indefinitely so I need to
> find a way to kill the session.
>
> Reading RFC4028  the only
> response codes that should trigger a BYE are 408/481 so my aim is to
> intercept the 477 and alter it. Unfortunately I have been unable to
> intercept it within my OpenSIPs config as the messages don't hit
> failure_route or similar. Am I missing a trick? Is there somewhere where
> these can be intercepted or should I be looking into a solution on the
> media server instead?
>
> Many thanks,
>
> Callum
>
>
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Re: [OpenSIPS-Users] [Crash Report] Weird crash with drouting/tls_mgm/usrloc/db_postgresql

[Johan De Clercq]

I believe that 1579 was fixed by volga629 by reducing the number of workers
in opensips.cfg.

Op di 11 aug. 2020 om 10:39 schreef Adrien Martin :

> Hello,
>
> As i am adding some TLS gateways (drouting) to our configuration some
> crashes happen.
> The crash dump seems to show db_postgresql is reading an answer that does
> not match the current query (usrloc).
>
> I don't really know how to find what causes this.
> Does anyone have an idea how to progress on this? How to extract more from
> the crash dump/what to do to find the problem?
>
> Does anyone have experienced this problem?
> NB: https://github.com/OpenSIPS/opensips/issues/1579 seems similar but
> how the patch interact with the issue is not obvious to me.
>
> More information is present here:
> https://github.com/OpenSIPS/opensips/issues/2161.
>
> Regards,
> --
> Adrien Martin
>
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Re: [OpenSIPS-Users] 3.1 - access Path and Via values from REGISTER

[Johan De Clercq]

t_relay to the socket on which you are listening.

Op ma 3 aug. 2020 om 12:21 schreef Mark Allen :

> > If you want to see it, loopback the message.
>
> Thanks. How do I do that?
>
> On Mon, 3 Aug 2020 at 11:02, Johan De Clercq  wrote:
>
>> I think that you are right.
>> If you want to see it, loopback the message.
>>
>> Op ma 3 aug. 2020 om 11:16 schreef Mark Allen :
>>
>>> Would I be right in thinking that the reason I cannot immediately see
>>> the "path" value added by the add_path_received() call is because of how
>>> the lumps system works - i.e.
>>>
>>>   "lumps are stored in a list, and are only applied
>>>after the OpenSIPS script is fully executed and
>>>before the SIP message is relayed. Because of
>>>this, changes done on a SIP message are not
>>>immediately reflected on the SIP message upon
>>>further inspection ( eg. Adding a new header from
>>>the script and then checking for the header's
>>>existence )."
>>>
>>> From: https://www.opensips.org/Documentation/Development-Manual
>>>
>>> On Thu, 30 Jul 2020 at 16:24, Mark Allen  wrote:
>>>
>>>> Seeking to find a workaround for the problem I'm having with WebRTC and
>>>> AOR throttling (
>>>> http://lists.opensips.org/pipermail/users/2020-July/043542.html) I
>>>> want to access the values of the "Via" and "Path" headers that are being
>>>> passed to the registrar.
>>>>
>>>> Using sngrep on the OpenSIPS server I can see the REGISTER includes
>>>> Path and Via headers. If I try to access them with  $(hdr(Path)[0])  or
>>>>  $(hdr(Via)[0])  I get nothing, but I can access the values of all the
>>>> other headers without any problem - e.g.  $(hdr(Authorization)[0]). What am
>>>> I missing or is there another way to get the info used in the Via
>>>> and particularly in creating the Path header values???
>>>>
>>>>   add_path_received();
>>>>   xlog(" $(hdr(Path)[0]) !!");
>>>>   $avp(attr) = $(hdr(Path)[0]);
>>>>   mid_registrar_save("location","p0v");
>>>>
>>>> The code gives an "attr" value in the "location" table of "NULL"
>>>>
>>>> /var/log/syslog shows:
>>>>
>>>>   Jul 30 16:13:17  /usr/sbin/opensips[27423]:  
>>>> !!
>>>>
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Re: [OpenSIPS-Users] 3.1 - access Path and Via values from REGISTER

[Johan De Clercq]

I think that you are right.
If you want to see it, loopback the message.

Op ma 3 aug. 2020 om 11:16 schreef Mark Allen :

> Would I be right in thinking that the reason I cannot immediately see the
> "path" value added by the add_path_received() call is because of how the
> lumps system works - i.e.
>
>   "lumps are stored in a list, and are only applied
>after the OpenSIPS script is fully executed and
>before the SIP message is relayed. Because of
>this, changes done on a SIP message are not
>immediately reflected on the SIP message upon
>further inspection ( eg. Adding a new header from
>the script and then checking for the header's
>existence )."
>
> From: https://www.opensips.org/Documentation/Development-Manual
>
> On Thu, 30 Jul 2020 at 16:24, Mark Allen  wrote:
>
>> Seeking to find a workaround for the problem I'm having with WebRTC and
>> AOR throttling (
>> http://lists.opensips.org/pipermail/users/2020-July/043542.html) I want
>> to access the values of the "Via" and "Path" headers that are being passed
>> to the registrar.
>>
>> Using sngrep on the OpenSIPS server I can see the REGISTER includes Path
>> and Via headers. If I try to access them with  $(hdr(Path)[0])  or
>>  $(hdr(Via)[0])  I get nothing, but I can access the values of all the
>> other headers without any problem - e.g.  $(hdr(Authorization)[0]). What am
>> I missing or is there another way to get the info used in the Via
>> and particularly in creating the Path header values???
>>
>>   add_path_received();
>>   xlog(" $(hdr(Path)[0]) !!");
>>   $avp(attr) = $(hdr(Path)[0]);
>>   mid_registrar_save("location","p0v");
>>
>> The code gives an "attr" value in the "location" table of "NULL"
>>
>> /var/log/syslog shows:
>>
>>   Jul 30 16:13:17  /usr/sbin/opensips[27423]:  
>> !!
>>
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Re: [OpenSIPS-Users] Opensips MS Teams lost BYE

[Johan De Clercq]

i dont see how you can  handle moh and attended transfer with opensips only.

On Mon, Jul 27, 2020, 11:31 Chris Johnson  wrote:

> We were only able to remove one of the fix_nated_contact lines before it
> broke more than it did fix.
>
>
>
> We’re currently seeking support from Opensips directly to get it all
> working correctly.
>
>
>
> *From:* Mark Miranda 
> *Sent:* 24 July 2020 21:21
> *To:* john.qu...@smartvox.co.uk; OpenSIPS users mailling list <
> users@lists.opensips.org>
> *Cc:* Chris Johnson 
> *Subject:* Re: [OpenSIPS-Users] Opensips MS Teams lost BYE
>
>
>
> CAUTION: This email originated from outside of the organisation. Do not
> click links or open attachments unless you recognise the sender and know
> the content is safe.
>
> I have basically the same problem where the BYE packet does not seem to be
> routing properly, however I am not using any of the nat functions.
>
> > On Jul 6, 2020, at 3:46 AM, John Quick 
> wrote:
> >
> > Chris,
> >
> > You must not call fix_nated_contact() for any SIP requests or SIP
> responses
> > coming from MS Teams.
> > If you do, it will cause exactly the problems you describe for
> loose-routed
> > sequential requests such as BYE.
> >
> > Just to explain: fix_nated_contact() should only be used when the
> messages
> > come via a NAT router. It must not be used for messages coming from a
> Proxy
> > (assuming the Proxy adds a Record-Route header, as happens with Teams),
> even
> > though both cases may appear to need fixing when you use nat_uac_test().
> >
> > John Quick
> > Smartvox Limited
> > Web: www.smartvox.co.uk
> >
> >> I've managed to get the PBX to send the BYE to the SBC and it is now
> > forwarding on to MS (although MS are not sending a reply but we have
> opened
> > a case for this).
> >
> >> We are struggling to get the BYE packet to go back to MS Teams when the
> > call is initiated by Teams.
> >
> >
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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[OpenSIPS-Users] bugs on smpp.

[Johan De Clercq]

Hi,

a few months ago, volga629 and myself added bugs and a feature request for
proto_smpp.
Now, after 30 days these issues are closed.

Is there any chance that these can be picked up ?

wkr,
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Re: [OpenSIPS-Users] opensips-cp for 3.1

[Johan De Clercq]

thanks Razvan,
btw, I will be starting on push notifications also.



Op ma 27 jul. 2020 om 09:32 schreef Răzvan Crainea :

> Hi, Johan!
>
> We haven't got a chance to tune OpenSIPS CP for 3.1, although we need to
> do this ASAP.
> Nevertheless, there were no major changes between 3.0 and 3.1, therefore
> you should be able to use 3.1.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 7/24/20 4:53 PM, johan wrote:
> >
> > Hi,
> >
> >
> > is it okay to use opensips-cp 3.0 for opensips 3.1 ?
> >
> >
> > wkr,
> >
> >
> > ___
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Re: [OpenSIPS-Users] opensips-cp for 3.1

[Johan De Clercq]

Thanks sharad

Outlook voor iOS downloaden

Van: Users  namens Sharad Kumar via Users 

Verzonden: Saturday, July 25, 2020 6:21:13 PM
Aan: users@lists.opensips.org 
Onderwerp: Re: [OpenSIPS-Users] opensips-cp for 3.1

Hi Johan,


Absolutely, You can use opensips-cp 3.0 for opensips 3.1





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Re: [OpenSIPS-Users] Make opensips register to VoIP Provider as a trunk with auth

[Johan De Clercq]

You need registrar and you need to put your credentials in the registrar table.

Outlook voor iOS downloaden

Van: Users  namens Sharad Kumar via Users 

Verzonden: Friday, July 10, 2020 2:56:06 AM
Aan: users@lists.opensips.org 
Onderwerp: [OpenSIPS-Users] Make opensips register to VoIP Provider as a trunk 
with auth

Hey guys,

I was just curious that can we make OpenSIPS to register as a SIP endpoint to 
VoIP provider like Twillio. We already using IP based auth in OpenSIPS to 
accept calls but one of our VOIP Provider wants our OpenSIPS SBC to register 
with them as a SIP trunk to get calls.
Does anyone got make OpenSIPS register to provider before using AUTH or 
Registrar module ?

Thank you



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Re: [OpenSIPS-Users] TLS handshake failure

[Johan De Clercq]

Which means that in b2b implemenation, you need to change contact header to
include fqdn.

wkr,

Op di 7 jul. 2020 om 10:36 schreef Gregory Massel :

> Thank you Alexey.
>
> Your suggestions solved the problem!
>
> On 2020-07-07 8:15 am, Alexey Vasilyev wrote:
> > Hi Gregory,
> >
> > Starting from 01.07.2020 Microsoft requires Route headers with SBC_FQDN
> in
> > SIP replies too. Instead of simple record_route() for SIP request from MS
> > servers OpenSIPS should add FQDN.
> >
> > I've updated the article. Check it.
> > https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/
> >
> >
> >
> > -
> > ---
> > Alexey Vasilyev
> > --
> > Sent from:
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
> >
> > ___
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> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] Push notifications server

[Johan De Clercq]

Adrian,
Where can I download this ?

Outlook voor iOS downloaden

Van: Users  namens Adrian Georgescu 

Verzonden: woensdag, juni 17, 2020 5:53 PM
Aan: OpenSIPS users mailling list
Onderwerp: [OpenSIPS-Users] Push notifications server

Hello,

We just made public a mobile push notification server that may help in various 
scenarios.

An integration guide for OpenSIPS is available in the source code.

https://ag-projects.com/news/sylk-pushserver/

Regards,
Adrian



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Re: [OpenSIPS-Users] Write acc_extra data to separate table

[Johan De Clercq]

If I am not wrong, you can specify the accounting table and db in the
module params.
Hence make a copy of the table, insert in a new db, add your extra column
and adapt the module parameters.

wkr,

Op do 11 jun. 2020 om 12:18 schreef Mark Farmer :

> Hi everyone
>
> I have a couple of extra fields in my acc database table which is added by
> acc_extra. This causes issues when upgrading.
> I would prefer to have that data stored in a different database/table.
>
> Is there a nice way to do that?
> OpenSIPS 3.0 in this case.
>
> Many thanks
> Mark.
>
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Re: [OpenSIPS-Users] Maybe it's a bug

[Johan De Clercq]

Upwards seems best.

Outlook voor iOS downloaden

Van: Users  namens Liviu Chircu 

Verzonden: Tuesday, June 9, 2020 8:58:24 PM
Aan: OpenSIPS users mailling list 
Onderwerp: Re: [OpenSIPS-Users] Maybe it's a bug

On 09.06.2020 15:43, Saint Michael wrote:
> I talked to Vlad, who I believe wrote the code, and he does not think
> it is a bug and I should use the ms and not the seconds. But thousands
> of businessmen will not spot this and thus their billing will never
> match the carrier, and they will lose money. If anybody thinks for a
> second that a call with a 200 OK will be free, is dreaming. Not in
> America.

Hi, SM!

Opinion #1: I doubt that anyone who is serious about their billing &
revenue (e.g. your nitpicky carrier) would leave to randomness the
answer to the most basic question of: "does our platform correctly bill
each call?".  No disrespect here, just maybe highlighting the fact that
your platform could benefit from a bit more testing.

Opinion #2: we could definitely change the default of the
second-accurate precision to be _greedy_ instead of _generous_. I bet
most people (myself included) would be more happy with a ceil() [1]
behavior instead of a trunc() [2] one.  That is: round _upwards_, not
_downwards_.  More opinions would be useful here!

Best regards,

[1]: man ceil
[2]: man trunc

--
Liviu Chircu
www.twitter.com/liviuchircu | 
www.opensips-solutions.com


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Re: [OpenSIPS-Users] ACK Routing Issue

[Johan De Clercq]

Call record route on initial invite.

Outlook voor iOS downloaden

Van: Users  namens Mark Farmer 

Verzonden: Friday, June 5, 2020 3:05:42 PM
Aan: OpenSIPS users mailling list 
Onderwerp: Re: [OpenSIPS-Users] ACK Routing Issue

Thanks Diptesh

I'm using topology hiding, so now I have this:

if (has_totag()) {
xlog("CUSTOM_LOG: in-dialog $rm has message flags: $mf and 
branch flags: $bf");

#Set correct SIP User-Agent Header
if (remove_hf("User-Agent")) {
xlog("CUSTOM_LOG: Setting SIP User-Agent on In-Dialog Request");
insert_hf("User-Agent: OpenSIPS\r\n");
}

if (!topology_hiding_match() ) {
xlog("CUSTOM_LOG: cannot match request to a dialog \n");
send_reply(404,"Not found");
}

# handle hop-by-hop ACK (no routing required)
if ( is_method("ACK") && t_check_trans() ) {
xlog("CUSTOM_LOG: ACK detected with valid transaction - 
t_relay");
t_relay();
exit;
}

I don't see a 404 going out so I think topology_hiding_match is working.
But it tries to send the ACK to itself on it's private interface (I have 
mhomed=1).

ACK 
sip:+44XX@10.150.50.72;did=e07.595f3776
 SIP/2.0
Via: SIP/2.0/UDP PUB.LIC.IP.ADDR:5060;branch=z9hG4bKc219.d1f5b08.2
From: ;tag=gK0c801c8d
To: 
mailto:sip%3A%2B44XX@147.152.17.42>>;tag=3800350621-1224267434
Call-ID: 
543691539-3800350621-1514620...@sbc-uk-bs13b.uk.sdin.bt.net
CSeq: 202841 ACK
Max-Forwards: 69
Content-Length: 0

Best regards
Mark.


On Fri, 5 Jun 2020 at 13:03, Diptesh Patel 
mailto:diptesh.pa...@ecosmob.com>> wrote:
Hello Mark,

Are you using Topology Hiding or Loose Routing?

If you are using Topology Hiding then you need to match the topology using 
topology_hiding_match() first.

It is great if you can share SIP packets.

Thanks & Regards
Diptesh Patel
Software Developer
Ecosmob Technologies Ltd,
Ahmedabad
Mo:+919898962659


On Fri, Jun 5, 2020 at 5:00 PM Mark Farmer 
mailto:farm...@gmail.com>> wrote:
Hi everyone

I've upgraded an OpenSIPS box to 3.1 and am now seeing an issue with ACK's 
trying to route to an incorrect IP - in this case our own advertised IP.

I think I'm right in saying that PRACK's & ACK's are treated equally and should 
route in the same manner? However, PRACK's are routing correctly.

I have this:

if (has_totag()) {
---

# handle hop-by-hop ACK (no routing required)
#if ( is_method("ACK") && t_check_trans() ) {
if (is_method("ACK")) {
t_relay();
exit;
}
---

Thanks for any ideas!
Mark.


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Re: [OpenSIPS-Users] OpenSIPS 3.1 release - the final countdown

[Johan De Clercq]

Then this is the first feature request for 3.2 :-).

Outlook voor iOS<https://aka.ms/o0ukef> downloaden

Van: Bogdan-Andrei Iancu 
Verzonden: Thursday, May 14, 2020 2:58:15 PM
Aan: OpenSIPS users mailling list ; Johan De Clercq 
; developensips 
Onderwerp: Re: [OpenSIPS-Users] OpenSIPS 3.1 release - the final countdown

Johan,

It is the same b2b engine, but instead of controlling  it via the XML file, you 
will control it via a script route.

Regards,

Bogdan-Andrei Iancu

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



On 5/14/20 1:52 PM, Johan De Clercq wrote:
Bogdan,

Does b2b script support means that we can handle refer fully in the script I.e. 
send 202 and generate invite without b2bua instance?

Outlook voor iOS<https://aka.ms/o0ukef> downloaden

Van: Users 
<mailto:users-boun...@lists.opensips.org> 
namens Bogdan-Andrei Iancu <mailto:bog...@opensips.org>
Verzonden: Thursday, May 14, 2020 11:01:18 AM
Aan: users@lists.opensips.org<mailto:users@lists.opensips.org> 
<mailto:users@lists.opensips.org>; developensips 
<mailto:de...@lists.opensips.org>
Onderwerp: [OpenSIPS-Users] OpenSIPS 3.1 release - the final countdown

Hi all,

We planned an ambitious roadmap [1] for OpenSIPS 3.1, but we were even
more ambitious by trying to complete it.  It was a long way,
brainstorming, designing, implementing or extended an awesome feature
set that will make OpenSIPS an appealing platform for Class 5 services.

But we are almost done and the date is set:
 27th of May, the date of the 3.1 beta release

To check the status, visit anytime the roadmap page and scroll at the
end for the status table - yes, we do update it in realtime ;)

[1] https://www.opensips.org/Development/Opensips-3-1-Planning

Best regards,

--
Bogdan-Andrei Iancu

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


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Re: [OpenSIPS-Users] OpenSIPS 3.1 release - the final countdown

[Johan De Clercq]

Bogdan,

Does b2b script support means that we can handle refer fully in the script I.e. 
send 202 and generate invite without b2bua instance?

Outlook voor iOS downloaden

Van: Users  namens Bogdan-Andrei Iancu 

Verzonden: Thursday, May 14, 2020 11:01:18 AM
Aan: users@lists.opensips.org ; developensips 

Onderwerp: [OpenSIPS-Users] OpenSIPS 3.1 release - the final countdown

Hi all,

We planned an ambitious roadmap [1] for OpenSIPS 3.1, but we were even
more ambitious by trying to complete it.  It was a long way,
brainstorming, designing, implementing or extended an awesome feature
set that will make OpenSIPS an appealing platform for Class 5 services.

But we are almost done and the date is set:
 27th of May, the date of the 3.1 beta release

To check the status, visit anytime the roadmap page and scroll at the
end for the status table - yes, we do update it in realtime ;)

[1] https://www.opensips.org/Development/Opensips-3-1-Planning

Best regards,

--
Bogdan-Andrei Iancu

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


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Re: [OpenSIPS-Users] MS Teams REFER transfer on hold.

[Johan De Clercq]

Can you send a trace (pcap)?

Outlook voor iOS downloaden

Van: Users  namens Yuriy Nasida 

Verzonden: Monday, May 11, 2020 8:50:44 AM
Aan: OpenSIPS users mailling list 
Onderwerp: Re: [OpenSIPS-Users] MS Teams REFER transfer on hold.

Guys,

Hold/Unhold, blind transfers is fine without REFER support
But... Please note that attended  (consult) transfer will not work.
I did many tests and MS does 3th call to connect  Transferee and Transfer 
Target. Not sure why. I tried same things from regular softphone - and it works 
great without 3th call.



On Sat, 9 May 2020 at 00:02, Slava Bendersky via Users 
mailto:users@lists.opensips.org>> wrote:
Hello Johan,
Thank you for reply and help.
Based on MS Teams   Allow header will disable REFER and MS Teams will send 
invite in dialog.
The following code  snippet on 200 OK reply  is resolve the issue.

$var(refer-allow) = "INVITE, ACK, CANCEL, BYE, INFO, 
NOTIFY, PRACK, UPDATE, OPTIONS";
remove_hf("Allow");
insert_hf("Allow: $var(refer-allow)\r\n");



volga629

From: "johan" mailto:jo...@democon.be>>
To: "OpenSIPS users mailling list" 
mailto:users@lists.opensips.org>>
Sent: Friday, May 8, 2020 9:58:56 AM
Subject: Re: [OpenSIPS-Users] MS Teams REFER transfer on hold.


try with replying that you don't support refer.

On 8/05/2020 11:19, Slava Bendersky via Users wrote:
Hello Everyone,
I am trying put tougher transfer feature with in MS Teams.
And based on my research and testing right now is not possible with opensips.
MS Teams are sending on hold  in case of transfer and onhold as a=inactive.
Major problem they require encode/decode params in REFER-TO REFER-BY.

MS Teams expect observe REFER with 202 and decode params  with future action.

Here official requirements doc.

https://docs.microsoft.com/en-us/microsoftteams/direct-routing-protocols-sip#call-transfer

Any suggestion welcome.

volga629.





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Re: [OpenSIPS-Users] Something odd with TLS Memory

[Johan De Clercq]

then you are good.

Op di 12 mei 2020 om 12:16 schreef James Hogbin :

> In that case I’m missing something as all SIP messages are being presented
> unencrypted in my Homer & SIPTRACE in the opensips-cp
>
> Well for the use case I’m trying to implement anyway
>
> MSTeams <-> Opensips <-> Freeswitch
>
> With Opensips feeding Homer
>
> J/.
>
>
>
> *James Hogbin*
> *Director*
> [image: IP Sentinel Logo] <http://ip-sentinel.com>
> t. 020 3011 4150 <+442030114150>
> m. +44 7786910895
> w. https://www.ip-sentinel.com
>
> On 12 May 2020, at 11:02, Johan De Clercq  wrote:
>
> My two cents : upgrade, if you have a trace of the TLS part then your life
> becomes much easier.
>
> Op di 12 mei 2020 om 11:59 schreef James Hogbin :
>
>> Thank you.
>>
>> Given I’m more focused on getting Teams to work and I can still get the
>> HEP trace without it I’ll wait.
>>
>> James
>>
>> *James Hogbin*
>> * Director*
>> <120051210572101339.png> <http://ip-sentinel.com/>
>> t. +44 (0)20 3011 4150 <+442030114150>
>> m. +44 7786910895
>> w. https://www.ip-sentinel.com
>>
>>
>> > On 12 May 2020, at 10:54, Liviu Chircu  wrote:
>> >
>> > On 12.05.2020 12:51, James Hogbin wrote:
>> >> version: opensips 3.0.2 (x86_64/linux)
>> >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>> >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>> >> poll method support: poll, epoll, sigio_rt, select.
>> >> main.c compiled on with gcc 7
>> >
>> > OK, so you don't have the fix, which was made in February. Please
>> re-install via 3.0 nightly packages if you want to fix the issue
>> immediately, otherwise wait for the 3.0.3 minor release.
>> >
>> > Best regards,
>> >
>> > --
>> > Liviu Chircu
>> > www.twitter.com/liviuchircu | www.opensips-solutions.com
>> >
>> >
>> > ___
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>>
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