Re: [OpenSIPS-Users] Certification

2015-09-21 Thread Flavio Goncalves
Hello Stas,

The enrollment key for the retake is #4certificationexam#. Instructions for
the test will be sent at the test time.

Best regards,



Flavio E. Goncalves
OpenSIPS Training

2015-09-08 19:30 GMT-03:00 Stas Kobzar :

> Hello Flavio,
>
> I see that it is possible to enroll now for the next OCP exam on
> ebootcamp.opensips.org web-site.
>
> About a year ago I have failed my OCP exam.
> Do I have to pay to re-try now?
>
> I have passed a theoretical part and failed only practical. Do I have to
> pass both now or only practical?
>
> Thank you!
>
> On Wed, Sep 2, 2015 at 11:23 AM, Flavio Goncalves 
> wrote:
>
>> Hello Stas,
>>
>> We are planning the last test for OpenSIPS 1.x for October 2nd. We will
>> open the registration until friday this week.
>>
>> Best regards,
>>
>> Flavio E. Goncalves
>>
>> 2015-09-02 11:26 GMT-03:00 Stas Kobzar :
>>
>>> Hello,
>>>
>>> I hope this is a good list to post my question.
>>>
>>> I am interested in OpenSIPS Certified Professional.
>>> Is there any upcoming certification exam planned in nearest future?
>>>
>>> Thank you!
>>>
>>> --
>>>
>>> Stas Kobzar
>>>
>>> Developeur VoIP / VoIP Developer
>>>
>>> ___
>>>
>>> Modulis­.ca Inc.
>>>
>>> # Bureau / Office: 514-284-2020 x 246
>>>
>>> Email: s tas.kob...@modulis.ca
>>>
>>> https://www.modulis.com
>>>
>>> 
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
>
> Stas Kobzar
>
> Developeur VoIP / VoIP Developer
>
> ___
>
> Modulis­.ca Inc.
>
> # Bureau / Office: 514-284-2020 x 246
>
> Email: s tas.kob...@modulis.ca
>
> https://www.modulis.com
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


Re: [OpenSIPS-Users] reinviting to a recording server

2015-09-21 Thread Tito Cumpen
Eric,


I am going to be using the dialogic XMS. I believe it handles SAVPF(DTLS).
The request to record will be triggered by an api. Below is a diagram of
what I intend to do.


Bogdan,

I'd like to know if I can trigger a reinvite via the MI_http interface
which will enact a b2bua scenario with the intention to move both legs to
the media server. Below is a diagram. One detail I'd like to point out is
the blank reinvite needs to source from opensips as it carries all the
headers. Please advise if this is possible or if I can do anything aside
from using a b2bua scenario.






[image: Inline image 1]




On Wed, Sep 9, 2015 at 6:09 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Tito,
>
> A SIP call can have only 2 end-points. What is not clear for me is: after
> inserting the media server, what is the final configuration in terms of
> who's talking to who? Still A talks to B, but media server is recording ?
> or A talks to media server (like VM) and B drops out ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 08.09.2015 21:27, Tito Cumpen wrote:
>
> Bogdan,
>
> Thanks for your reply and questions. Currently call flows are using ICE
> and rtpengine as a turn relay and so there's nothing in between . In the
> case I get a request to begin recording I'd like to move the active call to
> a media server that bridges the call making it appear seamless for the
> caller and callee. If I trigger a RE-INVITE to both A and B with the media
> server address this should work but I am not sure how I can use opensips to
> send a blank invite on behalf of both A and B utilizing the same call id to
> media server then utilizing the reply as the RE-INVITE to A and B. In
> essence putting the media server in between without forcing a hang up.
>
> Thanks,
> Tito
>
> On Mon, Sep 7, 2015 at 6:20 AM, Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Tito,
>>
>> Do you want to move on the call legs to the call recording server (like
>> to a VM or so) or while A talks to B, you want to have something in the
>> middle to record the call between those two parties ?
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 03.09.2015 01:13, Tito Cumpen wrote:
>>
>> Group,
>>
>> Has anyone had experience reinviting an ongoing session between two sip
>> clients to a sip capable media server for call recording purposes without
>> dropping the ongoing call? Is the best practice to use XML_RPCNG/fifo
>> command and have opensips interact as 3rd party call control. Or would the
>> 3rd party entity need to hijack the ongoing session  as pose as the remote
>> party. I have a requirement to record video and audio legs. The media
>> server is capable for recording these streams just need to find a way to do
>> this without dropping the call.
>>
>>
>> Thanks,
>> Tito
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


[OpenSIPS-Users] B2B Refer Scenario & SDP

2015-09-21 Thread David Sanders
Is there any way to use the refer B2B scenario, and still be able to modify
the SDP to account for NAT?

Specifically, inbound calls will have their 200 OK response entirely on the
B2B side, so the response is not able to be modified in the script.

I'd really like to find a solution, as the refer B2B scenario seems most
useful with an inbound call scenario (call comes in, transfer to a cell
phone or different internal network phone).

- David
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


Re: [OpenSIPS-Users] Table dialog remains empty even after calls. Some error in my config file? Others tables are good.

2015-09-21 Thread Rodrigo Pimenta Carvalho

Hi Razvan.


I have just checked the case and there is no issue in fact. I was 
misunderstanding the situation.

When the call ends, the OpenSIPS removes the record of the respective dialogue. 
That is why I was thinking that OpenSIPS wasn't saving data in the dialog 
table, as I was checking such table always after ending the call.


Now, I have to see whether there is a way of keeping records in the dialog 
table for some time, even after ending the calls. Because the information from 
dialog table will be useful for my application in moments after ending the 
calls.


I will study about ACC and Dialog. Let's see.


Thanks!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: users-boun...@lists.opensips.org  em nome 
de Razvan Crainea 
Enviado: quinta-feira, 17 de setembro de 2015 09:32
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Table dialog remains empty even after calls. Some 
error in my config file? Others tables are good.

Hi, Rodrigo!

Are you seeing any errors in your logs related to sqlite not being able to 
insert dialogs in your database?
Can you run a dlg_list while a call is on?

Best regards,

Razvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 09/17/2015 03:26 PM, Rodrigo Pimenta Carvalho wrote:



Hi.


Some days ago I got a help in this discussion list.

According to that help, the table that has information about dialled calls is 
acc.

So, I will read today about ACC and Dialog. I suspect that CDR information uses 
tables acc and dialog, due to the present fields (columns) in such tables.


Hopefully, I will see how to use such tables, making OpenSIPS record data 
there. Let's see...


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: sisc-requ...@listas.inatel.br 
 em nome 
de Rodrigo Pimenta Carvalho 
Enviado: quarta-feira, 16 de setembro de 2015 17:34
Para: OpenSIPS users mailling list
Assunto: [sisc] Table dialog remains empty even after calls. Some error in my 
config file? Others tables are good.


Dear OpenSIPS-Users,


I'm using OpenSIPS 2.2 with SQLite, from commit 
97ea216790a20e150d09314aeea360aa48a37b36.


I have noticed that my OpenSIPS is recording data only into the following 
tables:

- subscriber

- location

- missed_calls

-acc


I need make my OpenSIPS record data into table dialog too, as I need present 
the list of dialled calls in the GUI of my application. I intend to read 
dialled calls information from table dialog.

So, I suspect that maybe some configuration is wrong in my OpenSIPS environment.


In my opensips_residential.cfg file I have:





 DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "db_mode", 1)
modparam("dialog", "db_url", "sqlite:///usr/local/opensips_proxy/sqlite")
modparam("dialog", "default_timeout", 540)

---


### Routing Logic 


.

.

.


# account only INVITEs
if (is_method("INVITE")) {

# create dialog with timeout
if ( !create_dialog("B") ) {
send_reply("500","Internal Server Error");
exit;
}

setflag(ACC_DO); # do accounting
}

.

.

.


-


Is there some error in my configuration?

What should I investigate to eliminate the issue?


Any hint will be very helpful!


Thanks alot.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


Re: [OpenSIPS-Users] OpenSips-CP Not Working

2015-09-21 Thread Bogdan-Andrei Iancu

Hi,

be sure you have the MDB2 php pear installed on your system (see the 
Install Guide from the Documentation section at 
http://opensips-cp.sourceforge.net/)


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 19.09.2015 13:01, Muhammad Usman Shahid wrote:

I checked the logs and they have the following information

[Sat Sep 19 14:53:13.915427 2015] [:error] [pid 1928] [client 
127.0.0.1:47528 ] PHP Warning:  Creating 
default object from empty value in 
/var/www/opensips-cp/config/db.inc.php on line 25, referer: 
http://127.0.0.1/opensips-cp/web/


[Sat Sep 19 14:53:13.999544 2015] [:error] [pid 1928] [client 
127.0.0.1:47528 ] PHP Fatal error:  Call to 
undefined method MDB2_Error::setFetchMode() in 
/var/www/opensips-cp/web/db_connect.php on line 31, referer: 
http://127.0.0.1/opensips-cp/web/


Line 25 of db.inc.php is:

* $config->db_driver = "mysql";*

Line 31 of db_connect.php is:

*$link->setFetchMode(MDB2_FETCHMODE_ASSOC);*

I still don't understand what the problem is. Please help in solving this.

Regards


On Thu, Sep 17, 2015 at 12:06 AM, Giovanni Maruzzelli 
> wrote:


Check the web server logs

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18

On Sep 16, 2015 9:03 PM, "Muhammad Usman Shahid"
> wrote:

The following error occurs in Javascript console when I try to
login to Opensips-cp

POST http://192.168.1.15/opensips-cp/web/login.php 500
(Internal Server Error)

Please help

On Wed, Sep 16, 2015 at 10:38 PM, Muhammad Usman Shahid
> wrote:

Hello,

I am trying to run opensips-cp but after I input username
and password my browser opens "login.php" but nothing shows.

I am using the following tutorial.


http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration

Please help.

-- 
Muhammad Usman Shahid.





-- 
Muhammad Usman Shahid.


___
Users mailing list
Users@lists.opensips.org 
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org 
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




--
Muhammad Usman Shahid.


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


[OpenSIPS-Users] opensips 1.11.5 error opening new ssl connections with postgres

2015-09-21 Thread Iñigo Belamendia

Hi,

I sent this mail last week but for some reason I see it reached as an
empty message so I will add some new information here.

>From last monday (Sep 14) our OpenSIPS (1.11.5) dies after a restart.
The process starts but after few seconds (10") it goes down. That day we
did an apt-get update & apt-get upgrade and libpq5 package was upgraded.
A downgrade to previous version fixes the issue. All other apps against
Postgres in our organization work correctly. We have open a bug report
in Debian's bur reporting system [1].

In log we can see some access to database (Postgres), but after some
queries it refuses new queries execution. An error message appears in
log saying there's no authorization line in pg_hba.conf:

Sep 15 12:03:27 server01 opensips[17471]:
ERROR:db_postgres:db_postgres_new_connection: SSL error: called a
function you should not call#012FATAL:  no hay una l?nea en
pg_hba.conf para <<192.168.1.119>>, usuario <>, base de
datos <>, SSL inactivo#012

And it's true, theres no SSL-inactive auth line, but it is for SSL
(ip, username and password have been double checked). In attached log
file completed queries can be seen.

This issue happened in two different environments after a system
update (apt-get update && apt-get upgrade) and a service restart
(service opensips restart). Before each upgrade a disk snapshot is
taken but a restore from these snapshots (opensips and postgres)
doesn't fix the issue.

Queries executed [with psql] from console in OpenSIPS machine run ok.

Adding no-ssl auth into pg_hba.conf OpenSIPS starts ok.



[1] https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=799663



opensips.log.tgz
Description: application/compressed-tar
root@server01:~# uname -a
Linux server01 3.2.0-4-amd64 #1 SMP Debian 3.2.68-1+deb7u3 x86_64 GNU/Linux

root@server01:~# dpkg --list 'opensips*'
Deseado=Desconocido/Instalar/Eliminar/Purgar/Retener
| 
Estado=No/Instalado/Config-files/Desempaquetado/Medio-conf/Medio-inst/espera-disparo/pendiente-disparo
|/ Err?=(ninguno)/Requiere-reinst (Estado,Err: mayúsc.=malo)
||/ Nombre 
Versión Arquitectura
 Descripción
+++-==---=
ii  opensips   
1.11.5-1 amd64  
  very fast and configurable SIP server
un  opensips-b2bua-module  
   
  (no hay ninguna descripción disponible)
un  opensips-berkeley-module   
   
  (no hay ninguna descripción disponible)
un  opensips-carrierroute-module   
   
  (no hay ninguna descripción disponible)
un  opensips-console   
   
  (no hay ninguna descripción disponible)
un  opensips-cpl-module
   
  (no hay ninguna descripción disponible)
un  opensips-dbhttp-module 
   
  (no hay ninguna descripción disponible)
un  opensips-dialplan-module   
   
  (no hay ninguna descripción disponible)
un  opensips-geoip-module  
   
  (no hay ninguna descripción disponible)
un  opensips-http-modules  
   
  (no hay ninguna descripción disponible)
un  opensips-identity-module   
   
  (no hay ninguna descripción disponible)
un  opensips-jabber-module 
   
  (no hay ninguna descripción disponible)
un  opensips-json-module   
   
  (no hay ninguna descripción disponible)
ii  opensips-ldap-modules 

Re: [OpenSIPS-Users] Opensips Acc module .

2015-09-21 Thread Bogdan-Andrei Iancu

Hi Sasmita,

Unfortunately bad news - the db_cachedb can convert SQL to MongoDB only. 
Redis is not supported - my bad :(


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.09.2015 09:12, Sasmita Panda wrote:

Hi Bogdan ,

Now I am trying to compile  opensips-1.11 with db_cachedb module and 
cachedb_redis module .
As of now redis is working separately without this db_cachedb . When I 
am trying to combine both its throwing some

error .

Error while running opensips :

 NOTICE:db_cachedb:mod_init: initializing module db_cachedb ...
 NOTICE:cachedb_redis:mod_init: initializing module cachedb_redis ...
 WARNING:acc:mod_init: Integer flags are now deprecated! Use unique 
quoted strings!
 ERROR:db_cachedb:db_cachedb_query: The selected NoSQL driver cannot 
convert select queries

 ERROR:core:db_table_version: error in db_query
 ERROR:core:db_check_table_version: querying version for table acc
 ERROR:acc:acc_db_init: error during table version check
 ERROR:acc:mod_init: failed...did you load a database module?
 ERROR:core:init_mod: failed to initialize module acc
 ERROR:core:main: error while initializing modules
 NOTICE:cachedb_redis:destroy: destroy module cachedb_redis ...
 NOTICE:db_cachedb:destroy: destroy module db_cachedb ...


  I was trying to find the solution for this . But I saw you have 
suggested to someone that , there is some issue in
 redis module which cant . bellow is the link where you have suggested 
to use mangodb in place of redis .


http://comments.gmane.org/gmane.comp.voip.opensips.user/27693

 Can you please suggest me what can I do to fix this error . Waiting 
for your reply .


*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Thu, Sep 10, 2015 at 4:35 PM, Sasmita Panda > wrote:


Hi ,

 I saw your reply . But still my problem is not get solved .

  Firstly : For storing data in redis you replied to use  "
db_cachedb " module to convert sql data to non sql data and store
in redis . But this is module is available in opensips-1.11 but I
am using opensips-1.8 now and this version does not support
"db_cachedb" module  . I may use this process latter when I move
to the latest version of opensips . This is option is not relevant
to me now .

 Secondly : extra accounting option is also not suitable for
my query . let me explain you what extra accounting is doing and
what I want .

As of now accounting is is printing login in my syslog file
directly when a transaction completes
the logs look like :
*ACC: transaction answered:

timestamp=1441881036;method=INVITE;from_tag=99022274ae6e465eaaccfe6ac6a004d6;to_tag=e02c1a68;call_id=d56d3914a9a74aa6931f65dc0f11ccfd;code=200;reason=OK;*
*
*
   This is already happening . Now if I will use extra accounting
then that will append the extra log in this line .
For example : I added
   modparam("acc", "log_extra","uaA=$hdr(User-Agent)")
Now new  accounting log will look like bellow format
*ACC: transaction answered:

timestamp=1441881036;method=INVITE;from_tag=99022274ae6e465eaaccfe6ac6a004d6;to_tag=e02c1a68;call_id=d56d3914a9a74aa6931f65dc0f11ccfd;code=200;reason=OK;**uaA=WebAstra*
This adds extra parameter in the accounting message .  It wont
solve my problem .

  Now my requirement :
 Particularly, I just want the transaction timestamp .for
both INVITE and BYE , so that I can calculate the total time of a
call .
I want this particular timestamp in a variable or the
entire accounting information in a variable .
I can say . I will manage this in opensips.cfg file to print the
logs .

for example :  xlog("L_NOTICE", " Account $acc : callID $ci :
method $rm  \n");
In $acc , I want the accounting information so that I can store
this in redis by key:value pair .

   I hope now you will understand my requirement .Please  help me
if you have any solution .

Regards
Sasmita












*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Wed, Sep 9, 2015 at 9:37 PM, Bogdan-Andrei Iancu
> wrote:

Hi, see inlines.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 09.09.2015 18:09, Sasmita Panda wrote:


Hi ,

   Thank you so much for the early response .

 First of all I don't want to store the accounting
information in opensips data base . because it's getting
overloaded with lots of data .

 As you suggested , is this possible I can convert SQL db
to no SQL db in directly ,without storing the information 

[OpenSIPS-Users] B2B Top Hiding & Refer

2015-09-21 Thread David Sanders
Is it possible to run both the topology hiding and refer scenarios for B2B?
I've tried and get an error when establishing an outbound call when the ACK
should be sent.

Scenarios initialized:
b2b_init_request("top hiding");
b2b_init_request("refer");

Error:

INFO:b2b_logic:b2b_add_dlginfo: Dialog pair: [
6f614969-f34f542c@172.16.13.143] - [B2B.119.6282642]
ERROR:b2b_entities:b2b_send_request: State [3] not established, can not
send request ACK, [B2B.119.1393173]
ERROR:b2b_logic:b2b_logic_notify_request: Sending request failed
[B2B.119.1393173]
INFO:b2b_logic:b2b_add_dlginfo: Dialog pair: [
6f614969-f34f542c@172.16.13.143] - [B2B.119.1393173]


If this is not possible, are there any suggestions on doing topology hiding
and refer support at the same time?

- David
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


[OpenSIPS-Users] Fwd: opensips 1.11.5 error opening new ssl connections with postgres

2015-09-21 Thread Iñigo Belamendia
Sorry by resending messages, but they still reach empty or with few
lines


Hi,

I sent this mail last week but for some reason I see it reached as an
empty message so I will add some new information here.

Last monday (Sep 14) our OpenSIPS (1.11.5) went done after a restart.
The process starts but after few seconds (10") it goes down. That day we
did an apt-get update & apt-get upgrade and libpq5 package was upgraded.
A downgrade to previous version fixes the issue. All other apps against
Postgres in our organization work correctly. We have open a bug report
in Debian's bur reporting system [1].

In log we can see some access to database (Postgres), but after some
queries it refuses new queries execution. An error message appears in
log saying there's no authorization line in pg_hba.conf:

Sep 15 12:03:27 server01 opensips[17471]:
ERROR:db_postgres:db_postgres_new_connection: SSL error: called a
function you should not call#012FATAL:  no hay una l?nea en
pg_hba.conf para <<192.168.1.119>>, usuario <>, base de
datos <>, SSL inactivo#012

And it's true, theres no SSL-inactive auth line, but it is for SSL
(ip, username and password have been double checked). In attached log
file completed queries can be seen.

This issue happened in two different environments after a system
update (apt-get update && apt-get upgrade) and a service restart
(service opensips restart). Before each upgrade a disk snapshot is
taken but a restore from these snapshots (opensips and postgres)
doesn't fix the issue.

Queries executed [with psql] from console in OpenSIPS machine run ok.

Adding no-ssl auth into pg_hba.conf OpenSIPS starts ok.



[1] https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=799663






opensips.log.tgz
Description: application/compressed-tar


postgres.log.tgz
Description: application/compressed-tar
root@server01:~# uname -a
Linux server01 3.2.0-4-amd64 #1 SMP Debian 3.2.68-1+deb7u3 x86_64 GNU/Linux

root@server01:~# dpkg --list 'opensips*'
Deseado=Desconocido/Instalar/Eliminar/Purgar/Retener
| 
Estado=No/Instalado/Config-files/Desempaquetado/Medio-conf/Medio-inst/espera-disparo/pendiente-disparo
|/ Err?=(ninguno)/Requiere-reinst (Estado,Err: mayúsc.=malo)
||/ Nombre 
Versión Arquitectura
 Descripción
+++-==---=
ii  opensips   
1.11.5-1 amd64  
  very fast and configurable SIP server
un  opensips-b2bua-module  
   
  (no hay ninguna descripción disponible)
un  opensips-berkeley-module   
   
  (no hay ninguna descripción disponible)
un  opensips-carrierroute-module   
   
  (no hay ninguna descripción disponible)
un  opensips-console   
   
  (no hay ninguna descripción disponible)
un  opensips-cpl-module
   
  (no hay ninguna descripción disponible)
un  opensips-dbhttp-module 
   
  (no hay ninguna descripción disponible)
un  opensips-dialplan-module   
   
  (no hay ninguna descripción disponible)
un  opensips-geoip-module  
   
  (no hay ninguna descripción disponible)
un  opensips-http-modules  
   
  (no hay ninguna descripción disponible)
un  opensips-identity-module   
   
  (no hay ninguna descripción disponible)
un  opensips-jabber-module 
   
  (no hay ninguna descripción disponible)
un  opensips-json-module