Re: [OpenSIPS-Users] [OpenSIPS-News] OpenSIPS and FreeSWITCH Integration Training

[Owais Ahmad]

Hi Bogdan,

Will this training session be webcast?

Regards,
Owais

On Thu, Jul 28, 2016 at 2:51 AM, Bogdan-Andrei Iancu 
wrote:

> Hello,
>
> The detailed description of the training content can be found here :
>
> http://www.opensips.org/pub/events/2016-08-12_OpenSIPS-ClueCon_Chicago/ClueConTraining.pdf
>
> See you in Chicago,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 17.06.2016 17:35, Bogdan-Andrei Iancu wrote:
>
>> Hello,
>>
>> As part of ClueCon, we will run an one-day hands-on Training around
>> OpenSIPS and FreeSWITCH integration. The training follows the steps of
>> building a complete system featuring:
>> * OpenSIPS as a cluster front-end  - http://opensips.org
>> * FreeSWITCH PBX core system  - http://freeswitch.org/
>> * HOMER for SIP capturing  - http://sipcapture.org/
>> * CGRates as billing engine  - http://cgrates.org/
>> * SIP Fraud detection
>>
>> The training will be held on 12th of August 2016, as part of the ClueCon
>> event.
>>
>> Together with the training, we will run a new session of Design Clinics (
>> http://www.opensips.org/Community/Clinics)
>>
>> Details, registration (both training and clinics) and more are available
>> here:
>> http://www.opensips.org/events/Training-2016ClueCon.html
>>
>>
>> Best regards
>>
>>
>
> ___
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>
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Re: [OpenSIPS-Users] Is it possible to read parts of SDP? What is the module to do it?

[Bogdan-Andrei Iancu]

Hi Rodrigo,

Take a look at the {sdp} transformation, it might help you:
http://www.opensips.org/Documentation/Script-Tran-2-2#toc79

Regards,

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

On 27.07.2016 23:14, Rodrigo Pimenta Carvalho wrote:


Hi.


A SDP message is:


v=0
o=Z 0 0 IN IP4 192.168.21.40
s=Z
c=IN IP4 192.168.21.40
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


It can be in a SIP INVITE or in a SIP OK message.


How can I read the IP4 from there, in case of SIP INVITE or SIP OK, 
and get the value 192.168.21.40 for example?


Is there a module and function that provides such information in my 
script?



Any hint will be very helpful!


Best regards.



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


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Re: [OpenSIPS-Users] [OpenSIPS-News] OpenSIPS and FreeSWITCH Integration Training

[Bogdan-Andrei Iancu]

Hello,

The detailed description of the training content can be found here :
http://www.opensips.org/pub/events/2016-08-12_OpenSIPS-ClueCon_Chicago/ClueConTraining.pdf

See you in Chicago,

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

On 17.06.2016 17:35, Bogdan-Andrei Iancu wrote:

Hello,

As part of ClueCon, we will run an one-day hands-on Training around 
OpenSIPS and FreeSWITCH integration. The training follows the steps of 
building a complete system featuring:

* OpenSIPS as a cluster front-end  - http://opensips.org
* FreeSWITCH PBX core system  - http://freeswitch.org/
* HOMER for SIP capturing  - http://sipcapture.org/
* CGRates as billing engine  - http://cgrates.org/
* SIP Fraud detection

The training will be held on 12th of August 2016, as part of the 
ClueCon event.


Together with the training, we will run a new session of Design 
Clinics (http://www.opensips.org/Community/Clinics)


Details, registration (both training and clinics) and more are 
available here:

http://www.opensips.org/events/Training-2016ClueCon.html


Best regards




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[OpenSIPS-Users] Is it possible to read parts of SDP? What is the module to do it?

[Rodrigo Pimenta Carvalho]

Hi.


A SDP message is:


v=0
o=Z 0 0 IN IP4 192.168.21.40
s=Z
c=IN IP4 192.168.21.40
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


It can be in a SIP INVITE or in a SIP OK message.


How can I read the IP4 from there, in case of SIP INVITE or SIP OK, and get the 
value 192.168.21.40 for example?

Is there a module and function that provides such information in my script?


Any hint will be very helpful!


Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] Get concurrent calls from sip server.

[Cesar Alberto Rodriguez Fierro]

There was a wrong configuration of the  "dialog.so" module.

I will try to parse the output in order to obtain the current calls.

 Thanks for the help.

On Wed, Jul 27, 2016 at 1:20 PM, Carlos Eduardo  wrote:

> Cesar,
>
> Are you using the dialog module in your script?
>
> This MI command will only return a valid value if the dialog module is
> loaded (loadmodule "dialog.so") and if the dialogs are criated durign the
> script processing (create_dialog() command).
>
> 2016-07-27 16:12 GMT-03:00 Daniel Zanutti :
>
>> On my sample, you should run:
>>
>> opensipsctl fifo profile_get_size inbound
>>
>>
>> Using dlg_list you should get something like this:
>> # opensipsctl fifo dlg_list
>> dialog::  hash=84:1411689852
>> state:: 4
>> user_flags:: 0
>> timestart:: 1469646635
>> datestart:: 2016-07-27 16:10:35
>> timeout:: 1469653835
>> dateout:: 2016-07-27 18:10:35
>> ...
>> dialog::  hash=289:324429409
>> state:: 2
>> user_flags:: 0
>> timestart:: 0
>> timeout:: 0
>> ...
>> dialog::  hash=640:1695114669
>> state:: 4
>> ...
>>
>> Check if modules are successfully loaded.
>>
>> Regards
>>
>> On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro <
>> c...@transtelco.net> wrote:
>>
>>>
>>> Thanks for your help.
>>>
>>> I trying to use  FIFO in order to  send requests to OpenSIPS, I have
>>> read some documentation about the  Dialog Module.  I guess using the 
>>> "opensipsctl
>>> fifo dlg_list" command can be useful to obtain the current calls, but I am
>>> not sure why the command is not available in my OpenSIPS version.   When I
>>> execute the command ./opensipsctl fifo version, I am getting the following
>>> information
>>> Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).
>>>
>>>
>>>
>>> On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <
>>> pime...@inatel.br> wrote:
>>>
 Now, thinking more about it, I would suggest you to put a SQL query in
 your proprietary software to query the OpenSIPS database directly.

 The table dialog will be always updated about current calls.


 Regards.


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


 --
 *De:* users-boun...@lists.opensips.org <
 users-boun...@lists.opensips.org> em nome de Rodrigo Pimenta Carvalho <
 pime...@inatel.br>
 *Enviado:* quarta-feira, 27 de julho de 2016 14:39
 *Para:* users@lists.opensips.org
 *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls from sip server.


 With FIFO you can send requests to OpenSIPS, for example from a
 proprietary software. So, if a request wants to execute a query with
 avpops, I guess FIFO will be useful.


 Regards.



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


 --
 *De:* users-boun...@lists.opensips.org <
 users-boun...@lists.opensips.org> em nome de Cesar Alberto Rodriguez
 Fierro 
 *Enviado:* quarta-feira, 27 de julho de 2016 14:29
 *Para:* users@lists.opensips.org
 *Assunto:* [OpenSIPS-Users] Get concurrent calls from sip server.

 Hi !

 I am currently working in a project related with display in real time
 the active calls of our VoIP traffic, I would like to get the active
 sip-calls from a Kamailio Sip Server (running opensips), is there any way
 to obtain this information.

 Best Regards.




 [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
 656-257-4112 |

 CONFIDENTIALITY NOTICE:  This communication is intended only for the
 use of the individual or entity to which it is addressed and may contain
 information that is privileged, confidential, and exempt from disclosure
 under applicable law.  If you are not the intended recipient of this
 information, you are notified that any use, dissemination, distribution, or
 copying of the communication is strictly prohibited.

 ___
 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
>>>
>>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> *Carlos E. Wagner*
> *Tecnólogo em Telecomunicações, OCP, dCAA*
>
> *Gnotel Telecom*
> *E-mail:* *kad...@gmail.com *
> *car...@gnotel.com.br *
> *Fone:* +55 48 

Re: [OpenSIPS-Users] Get concurrent calls from sip server.

[Rodrigo Pimenta Carvalho]

Hi.


In my case I used to run:


echo $'dlg_list\n' | xargs ./opensipsctl fifo > RespostasFIFO.txt   
or ./opensipsctl fifo dlg_list > RespostasFIFO.txt


The file RespostasFIFO.txt will be created automatically.

I my script I also have loadmodule "dialog.so".


Regards.


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



De: users-boun...@lists.opensips.org  em nome 
de Carlos Eduardo 
Enviado: quarta-feira, 27 de julho de 2016 16:20
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] Get concurrent calls from sip server.

Cesar,

Are you using the dialog module in your script?

This MI command will only return a valid value if the dialog module is loaded 
(loadmodule "dialog.so") and if the dialogs are criated durign the script 
processing (create_dialog() command).

2016-07-27 16:12 GMT-03:00 Daniel Zanutti 
>:
On my sample, you should run:

opensipsctl fifo profile_get_size inbound


Using dlg_list you should get something like this:
# opensipsctl fifo dlg_list
dialog::  hash=84:1411689852
state:: 4
user_flags:: 0
timestart:: 1469646635
datestart:: 2016-07-27 16:10:35
timeout:: 1469653835
dateout:: 2016-07-27 18:10:35
...
dialog::  hash=289:324429409
state:: 2
user_flags:: 0
timestart:: 0
timeout:: 0
...
dialog::  hash=640:1695114669
state:: 4
...

Check if modules are successfully loaded.

Regards

On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro 
> wrote:

Thanks for your help.

I trying to use  FIFO in order to  send requests to OpenSIPS, I have read some 
documentation about the  Dialog Module.  I guess using the "opensipsctl fifo 
dlg_list" command can be useful to obtain the current calls, but I am not sure 
why the command is not available in my OpenSIPS version.   When I execute the 
command ./opensipsctl fifo version, I am getting the following information
Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).



On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho 
> wrote:

Now, thinking more about it, I would suggest you to put a SQL query in your 
proprietary software to query the OpenSIPS database directly.

The table dialog will be always updated about current calls.


Regards.


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



De: users-boun...@lists.opensips.org 
> em 
nome de Rodrigo Pimenta Carvalho >
Enviado: quarta-feira, 27 de julho de 2016 14:39
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Get concurrent calls from sip server.


With FIFO you can send requests to OpenSIPS, for example from a proprietary 
software. So, if a request wants to execute a query with avpops, I guess FIFO 
will be useful.


Regards.



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



De: users-boun...@lists.opensips.org 
> em 
nome de Cesar Alberto Rodriguez Fierro 
>
Enviado: quarta-feira, 27 de julho de 2016 14:29
Para: users@lists.opensips.org
Assunto: [OpenSIPS-Users] Get concurrent calls from sip server.

Hi !

I am currently working in a project related with display in real time the 
active calls of our VoIP traffic, I would like to get the active sip-calls from 
a Kamailio Sip Server (running opensips), is there any way to obtain this 
information.

Best Regards.




[Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 
656-257-4112 |
[https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif]

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of the 
individual or entity to which it is addressed and may contain information that 
is privileged, confidential, and exempt from disclosure under applicable law.  
If you are not the intended recipient of this information, you are notified 
that any use, dissemination, distribution, or copying of the communication is 
strictly prohibited.

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Re: [OpenSIPS-Users] Get concurrent calls from sip server.

[Carlos Eduardo]

Cesar,

Are you using the dialog module in your script?

This MI command will only return a valid value if the dialog module is
loaded (loadmodule "dialog.so") and if the dialogs are criated durign the
script processing (create_dialog() command).

2016-07-27 16:12 GMT-03:00 Daniel Zanutti :

> On my sample, you should run:
>
> opensipsctl fifo profile_get_size inbound
>
>
> Using dlg_list you should get something like this:
> # opensipsctl fifo dlg_list
> dialog::  hash=84:1411689852
> state:: 4
> user_flags:: 0
> timestart:: 1469646635
> datestart:: 2016-07-27 16:10:35
> timeout:: 1469653835
> dateout:: 2016-07-27 18:10:35
> ...
> dialog::  hash=289:324429409
> state:: 2
> user_flags:: 0
> timestart:: 0
> timeout:: 0
> ...
> dialog::  hash=640:1695114669
> state:: 4
> ...
>
> Check if modules are successfully loaded.
>
> Regards
>
> On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro <
> c...@transtelco.net> wrote:
>
>>
>> Thanks for your help.
>>
>> I trying to use  FIFO in order to  send requests to OpenSIPS, I have
>> read some documentation about the  Dialog Module.  I guess using the 
>> "opensipsctl
>> fifo dlg_list" command can be useful to obtain the current calls, but I am
>> not sure why the command is not available in my OpenSIPS version.   When I
>> execute the command ./opensipsctl fifo version, I am getting the following
>> information
>> Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).
>>
>>
>>
>> On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <
>> pime...@inatel.br> wrote:
>>
>>> Now, thinking more about it, I would suggest you to put a SQL query in
>>> your proprietary software to query the OpenSIPS database directly.
>>>
>>> The table dialog will be always updated about current calls.
>>>
>>>
>>> Regards.
>>>
>>>
>>> RODRIGO PIMENTA CARVALHO
>>> Inatel Competence Center
>>> Software
>>> Ph: +55 35 3471 9200 RAMAL 979
>>>
>>>
>>> --
>>> *De:* users-boun...@lists.opensips.org 
>>> em nome de Rodrigo Pimenta Carvalho 
>>> *Enviado:* quarta-feira, 27 de julho de 2016 14:39
>>> *Para:* users@lists.opensips.org
>>> *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls from sip server.
>>>
>>>
>>> With FIFO you can send requests to OpenSIPS, for example from a
>>> proprietary software. So, if a request wants to execute a query with
>>> avpops, I guess FIFO will be useful.
>>>
>>>
>>> Regards.
>>>
>>>
>>>
>>> RODRIGO PIMENTA CARVALHO
>>> Inatel Competence Center
>>> Software
>>> Ph: +55 35 3471 9200 RAMAL 979
>>>
>>>
>>> --
>>> *De:* users-boun...@lists.opensips.org 
>>> em nome de Cesar Alberto Rodriguez Fierro 
>>> *Enviado:* quarta-feira, 27 de julho de 2016 14:29
>>> *Para:* users@lists.opensips.org
>>> *Assunto:* [OpenSIPS-Users] Get concurrent calls from sip server.
>>>
>>> Hi !
>>>
>>> I am currently working in a project related with display in real time
>>> the active calls of our VoIP traffic, I would like to get the active
>>> sip-calls from a Kamailio Sip Server (running opensips), is there any way
>>> to obtain this information.
>>>
>>> Best Regards.
>>>
>>>
>>>
>>>
>>> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
>>> 656-257-4112 |
>>>
>>> CONFIDENTIALITY NOTICE:  This communication is intended only for the use
>>> of the individual or entity to which it is addressed and may contain
>>> information that is privileged, confidential, and exempt from disclosure
>>> under applicable law.  If you are not the intended recipient of this
>>> information, you are notified that any use, dissemination, distribution, or
>>> copying of the communication is strictly prohibited.
>>>
>>> ___
>>> 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
>>
>>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
*Carlos E. Wagner*
*Tecnólogo em Telecomunicações, OCP, dCAA*

*Gnotel Telecom*
*E-mail:* *kad...@gmail.com *
*car...@gnotel.com.br *
*Fone:* +55 48 9981-0894
*Skype:* carlos.e.wagner
www.gnotel.com.br
www.blogdovoip.com
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Re: [OpenSIPS-Users] Get concurrent calls from sip server.

[Daniel Zanutti]

On my sample, you should run:

opensipsctl fifo profile_get_size inbound


Using dlg_list you should get something like this:
# opensipsctl fifo dlg_list
dialog::  hash=84:1411689852
state:: 4
user_flags:: 0
timestart:: 1469646635
datestart:: 2016-07-27 16:10:35
timeout:: 1469653835
dateout:: 2016-07-27 18:10:35
...
dialog::  hash=289:324429409
state:: 2
user_flags:: 0
timestart:: 0
timeout:: 0
...
dialog::  hash=640:1695114669
state:: 4
...

Check if modules are successfully loaded.

Regards

On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro <
c...@transtelco.net> wrote:

>
> Thanks for your help.
>
> I trying to use  FIFO in order to  send requests to OpenSIPS, I have read
> some documentation about the  Dialog Module.  I guess using the "opensipsctl
> fifo dlg_list" command can be useful to obtain the current calls, but I am
> not sure why the command is not available in my OpenSIPS version.   When I
> execute the command ./opensipsctl fifo version, I am getting the following
> information
> Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).
>
>
>
> On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <
> pime...@inatel.br> wrote:
>
>> Now, thinking more about it, I would suggest you to put a SQL query in
>> your proprietary software to query the OpenSIPS database directly.
>>
>> The table dialog will be always updated about current calls.
>>
>>
>> Regards.
>>
>>
>> RODRIGO PIMENTA CARVALHO
>> Inatel Competence Center
>> Software
>> Ph: +55 35 3471 9200 RAMAL 979
>>
>>
>> --
>> *De:* users-boun...@lists.opensips.org 
>> em nome de Rodrigo Pimenta Carvalho 
>> *Enviado:* quarta-feira, 27 de julho de 2016 14:39
>> *Para:* users@lists.opensips.org
>> *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls from sip server.
>>
>>
>> With FIFO you can send requests to OpenSIPS, for example from a
>> proprietary software. So, if a request wants to execute a query with
>> avpops, I guess FIFO will be useful.
>>
>>
>> Regards.
>>
>>
>>
>> RODRIGO PIMENTA CARVALHO
>> Inatel Competence Center
>> Software
>> Ph: +55 35 3471 9200 RAMAL 979
>>
>>
>> --
>> *De:* users-boun...@lists.opensips.org 
>> em nome de Cesar Alberto Rodriguez Fierro 
>> *Enviado:* quarta-feira, 27 de julho de 2016 14:29
>> *Para:* users@lists.opensips.org
>> *Assunto:* [OpenSIPS-Users] Get concurrent calls from sip server.
>>
>> Hi !
>>
>> I am currently working in a project related with display in real time the
>> active calls of our VoIP traffic, I would like to get the active sip-calls
>> from a Kamailio Sip Server (running opensips), is there any way to obtain
>> this information.
>>
>> Best Regards.
>>
>>
>>
>>
>> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
>> 656-257-4112 |
>>
>> CONFIDENTIALITY NOTICE:  This communication is intended only for the use
>> of the individual or entity to which it is addressed and may contain
>> information that is privileged, confidential, and exempt from disclosure
>> under applicable law.  If you are not the intended recipient of this
>> information, you are notified that any use, dissemination, distribution, or
>> copying of the communication is strictly prohibited.
>>
>> ___
>> 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
>
>
___
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Re: [OpenSIPS-Users] Get concurrent calls from sip server.

[Cesar Alberto Rodriguez Fierro]

Thanks for your help.

I trying to use  FIFO in order to  send requests to OpenSIPS, I have read
some documentation about the  Dialog Module.  I guess using the "opensipsctl
fifo dlg_list" command can be useful to obtain the current calls, but I am
not sure why the command is not available in my OpenSIPS version.   When I
execute the command ./opensipsctl fifo version, I am getting the following
information
Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).



On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <
pime...@inatel.br> wrote:

> Now, thinking more about it, I would suggest you to put a SQL query in
> your proprietary software to query the OpenSIPS database directly.
>
> The table dialog will be always updated about current calls.
>
>
> Regards.
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
>
> --
> *De:* users-boun...@lists.opensips.org 
> em nome de Rodrigo Pimenta Carvalho 
> *Enviado:* quarta-feira, 27 de julho de 2016 14:39
> *Para:* users@lists.opensips.org
> *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls from sip server.
>
>
> With FIFO you can send requests to OpenSIPS, for example from a
> proprietary software. So, if a request wants to execute a query with
> avpops, I guess FIFO will be useful.
>
>
> Regards.
>
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
>
> --
> *De:* users-boun...@lists.opensips.org 
> em nome de Cesar Alberto Rodriguez Fierro 
> *Enviado:* quarta-feira, 27 de julho de 2016 14:29
> *Para:* users@lists.opensips.org
> *Assunto:* [OpenSIPS-Users] Get concurrent calls from sip server.
>
> Hi !
>
> I am currently working in a project related with display in real time the
> active calls of our VoIP traffic, I would like to get the active sip-calls
> from a Kamailio Sip Server (running opensips), is there any way to obtain
> this information.
>
> Best Regards.
>
>
>
>
> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
> 656-257-4112 |
>
> CONFIDENTIALITY NOTICE:  This communication is intended only for the use
> of the individual or entity to which it is addressed and may contain
> information that is privileged, confidential, and exempt from disclosure
> under applicable law.  If you are not the intended recipient of this
> information, you are notified that any use, dissemination, distribution, or
> copying of the communication is strictly prohibited.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] Get concurrent calls from sip server.

[Rodrigo Pimenta Carvalho]

Now, thinking more about it, I would suggest you to put a SQL query in your 
proprietary software to query the OpenSIPS database directly.

The table dialog will be always updated about current calls.


Regards.


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



De: users-boun...@lists.opensips.org  em nome 
de Rodrigo Pimenta Carvalho 
Enviado: quarta-feira, 27 de julho de 2016 14:39
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Get concurrent calls from sip server.


With FIFO you can send requests to OpenSIPS, for example from a proprietary 
software. So, if a request wants to execute a query with avpops, I guess FIFO 
will be useful.


Regards.



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



De: users-boun...@lists.opensips.org  em nome 
de Cesar Alberto Rodriguez Fierro 
Enviado: quarta-feira, 27 de julho de 2016 14:29
Para: users@lists.opensips.org
Assunto: [OpenSIPS-Users] Get concurrent calls from sip server.

Hi !

I am currently working in a project related with display in real time the 
active calls of our VoIP traffic, I would like to get the active sip-calls from 
a Kamailio Sip Server (running opensips), is there any way to obtain this 
information.

Best Regards.




[Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 
656-257-4112 |
[https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif]

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of the 
individual or entity to which it is addressed and may contain information that 
is privileged, confidential, and exempt from disclosure under applicable law.  
If you are not the intended recipient of this information, you are notified 
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strictly prohibited.
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Re: [OpenSIPS-Users] Get concurrent calls from sip server.

[Rodrigo Pimenta Carvalho]

With FIFO you can send requests to OpenSIPS, for example from a proprietary 
software. So, if a request wants to execute a query with avpops, I guess FIFO 
will be useful.


Regards.



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



De: users-boun...@lists.opensips.org  em nome 
de Cesar Alberto Rodriguez Fierro 
Enviado: quarta-feira, 27 de julho de 2016 14:29
Para: users@lists.opensips.org
Assunto: [OpenSIPS-Users] Get concurrent calls from sip server.

Hi !

I am currently working in a project related with display in real time the 
active calls of our VoIP traffic, I would like to get the active sip-calls from 
a Kamailio Sip Server (running opensips), is there any way to obtain this 
information.

Best Regards.




[Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 
656-257-4112 |
[https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif]

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of the 
individual or entity to which it is addressed and may contain information that 
is privileged, confidential, and exempt from disclosure under applicable law.  
If you are not the intended recipient of this information, you are notified 
that any use, dissemination, distribution, or copying of the communication is 
strictly prohibited.
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Re: [OpenSIPS-Users] Get concurrent calls from sip server.

[Daniel Zanutti]

Hi Cesar

For realtime querying, you can use dialog profiles. Set a profile (for
example: inbound) on all calls like this:
set_dlg_profile("inbound");

Then read this using FIFO.

I just didn't get the "Kamailio Sip Server running Opensips".

Regards

On Wed, Jul 27, 2016 at 2:29 PM, Cesar Alberto Rodriguez Fierro <
c...@transtelco.net> wrote:

> Hi !
>
> I am currently working in a project related with display in real time the
> active calls of our VoIP traffic, I would like to get the active sip-calls
> from a Kamailio Sip Server (running opensips), is there any way to obtain
> this information.
>
> Best Regards.
>
>
>
>
> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
> 656-257-4112 |
>
> CONFIDENTIALITY NOTICE:  This communication is intended only for the use
> of the individual or entity to which it is addressed and may contain
> information that is privileged, confidential, and exempt from disclosure
> under applicable law.  If you are not the intended recipient of this
> information, you are notified that any use, dissemination, distribution, or
> copying of the communication is strictly prohibited.
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] Get concurrent calls from sip server.

[Rodrigo Pimenta Carvalho]

Hi.


Table dialog from the OpenSIPS database will contain information useful for 
your needing.

You will have to query the database to get data from such table and then handle 
data as you need.

For querying the database, see about 
http://www.opensips.org/html/docs/modules/2.2.x/avpops.html


Regards.

AVPops Module - 
OpenSIPS
www.opensips.org
AVPops (AVP-operations) modules implements a set of script functions which 
allow access and manipulation of user AVPs (preferences) and pseudo-variables.




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



De: users-boun...@lists.opensips.org  em nome 
de Cesar Alberto Rodriguez Fierro 
Enviado: quarta-feira, 27 de julho de 2016 14:29
Para: users@lists.opensips.org
Assunto: [OpenSIPS-Users] Get concurrent calls from sip server.

Hi !

I am currently working in a project related with display in real time the 
active calls of our VoIP traffic, I would like to get the active sip-calls from 
a Kamailio Sip Server (running opensips), is there any way to obtain this 
information.

Best Regards.




[Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 
656-257-4112 |
[https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif]

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of the 
individual or entity to which it is addressed and may contain information that 
is privileged, confidential, and exempt from disclosure under applicable law.  
If you are not the intended recipient of this information, you are notified 
that any use, dissemination, distribution, or copying of the communication is 
strictly prohibited.
___
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[OpenSIPS-Users] Get concurrent calls from sip server.

[Cesar Alberto Rodriguez Fierro]

Hi !

I am currently working in a project related with display in real time the
active calls of our VoIP traffic, I would like to get the active sip-calls
from a Kamailio Sip Server (running opensips), is there any way to obtain
this information.

Best Regards.




[image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 656-257-4112
 |

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of
the individual or entity to which it is addressed and may contain
information that is privileged, confidential, and exempt from disclosure
under applicable law.  If you are not the intended recipient of this
information, you are notified that any use, dissemination, distribution, or
copying of the communication is strictly prohibited.
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Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?

[Rodrigo Pimenta Carvalho]

Sorry. The rest of message:


1 - check the IP from SDP.

2 - If such IP is equal to the IP of the gateway, then:

we will fix the current IP, changing it to the OpenSIPS public IP.


Hopefully, it is easy to read such IP from SDP, using some function of OpenSIPS.


Could you comment about this decision, please?


Thank you very much!



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



De: Rodrigo Pimenta Carvalho
Enviado: quarta-feira, 27 de julho de 2016 14:04
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?


Hi.


It sounds good for who is using rtpproxy. In our case we are using direct media 
without rtpproxy.


As long as our OpenSIPS will be behind a NAT and such NAT is a gateway, we have 
decided to do the following:






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



De: users-boun...@lists.opensips.org  em nome 
de Johan De Clercq 
Enviado: quarta-feira, 27 de julho de 2016 11:57
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?

in the startup route, you should find  your pub ip using curl.
Put that in a var and then use that var in rtpproxy.

2016-07-27 15:07 GMT+02:00 Rodrigo Pimenta Carvalho 
>:

Dear OpenSIPS-users.



The function nat_uac_test() (from module NATHELPER) works very well and tells 
me whether a cliente to my OpenSIPS is behind a NAT.


But, for my specific network topology, is my OpenSIPS that is behind a NAT, 
from the client perspective. In this case I have to fix the SDP content that 
goes from OpenSIPS to the client, so that the client will be able to send its 
media to a public IP, when communicating to a peer in the same network domain 
of this SIP server.


How can I be sure that for a client perspective the OpenSIPS is behind a NAT? 
In another words, is there a way to the OpenSIPS determine if its client is in 
the same network or in a remote network?


--


P.S. I suspect that OpenSIPS should be used in a node with public IP to 
simplify our solution. But our customer asked us to put OpenSIPS in a 
residential device that will be always behind a NAT for some smartphones 
perspective and not behind a NAT for another home devices perspective.


---


Any hint will be very helpful!


Best regards.



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

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Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?

[Rodrigo Pimenta Carvalho]

Hi.


It sounds good for who is using rtpproxy. In our case we are using direct media 
without rtpproxy.


As long as our OpenSIPS will be behind a NAT and such NAT is a gateway, we have 
decided to do the following:






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



De: users-boun...@lists.opensips.org  em nome 
de Johan De Clercq 
Enviado: quarta-feira, 27 de julho de 2016 11:57
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?

in the startup route, you should find  your pub ip using curl.
Put that in a var and then use that var in rtpproxy.

2016-07-27 15:07 GMT+02:00 Rodrigo Pimenta Carvalho 
>:

Dear OpenSIPS-users.



The function nat_uac_test() (from module NATHELPER) works very well and tells 
me whether a cliente to my OpenSIPS is behind a NAT.


But, for my specific network topology, is my OpenSIPS that is behind a NAT, 
from the client perspective. In this case I have to fix the SDP content that 
goes from OpenSIPS to the client, so that the client will be able to send its 
media to a public IP, when communicating to a peer in the same network domain 
of this SIP server.


How can I be sure that for a client perspective the OpenSIPS is behind a NAT? 
In another words, is there a way to the OpenSIPS determine if its client is in 
the same network or in a remote network?


--


P.S. I suspect that OpenSIPS should be used in a node with public IP to 
simplify our solution. But our customer asked us to put OpenSIPS in a 
residential device that will be always behind a NAT for some smartphones 
perspective and not behind a NAT for another home devices perspective.


---


Any hint will be very helpful!


Best regards.



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

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Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

[Newlin, Ben]

I have identified that these crashes are occurring when the far end system is 
not returning the Record-Route headers in the 200 OK response. The headers are 
present in the 180 response, but not the 200 OK. I have reproduced the scenario 
using SIPp and captured a SIP trace: http://pastebin.com/ckKk3EhY

The crash occurs on receipt of the ACK request and attempt to match the dialog.

I also captured a BT for this scenario as well, in case anything specific in 
the trace made the issue easier to find: http://pastebin.com/cM3FhPiw

I am working with the other system to try to fix their behavior.

Ideally the Record-Route headers from previous replies could be used in this 
case to allow the call to succeed, but I don’t know if that is possible.

Thanks,

Ben Newlin

From: "Newlin, Ben" 
Date: Wednesday, July 27, 2016 at 9:44 AM
To: Bogdan-Andrei Iancu , OpenSIPS users mailling list 

Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Bogdan,

This is a different scenario than the other you responded to. As I said, we 
have two types of servers that work together. One is a load-balancer and runs 
as a proxy. It uses double Record-Route because it sends messages between 
public and private networks. Then we have our other servers using TH which 
receive those requests. We are not using TH and RR on the same server (although 
I would like to).

If validate_dialog() and fix_route_dialog() (and possibly loose_route()) should 
not be called when using TH, I believe the documentation should reference that. 
It states that match_dialog() must be used with TH, but does not indicate that 
the other functions should not be used or that the functionality won’t work. 
There is also no documentation of the incompatibility between RR and TH.

Either way, I ran a test where I removed all calls to loose_route(), 
validate_dialog(), and fix_route_dialog() from my script. The crash still 
occurred and the BT still pointed to fix_route_dialog() function. So it must be 
getting called from within Dialog module somewhere. That BT is here: 
http://pastebin.com/wu2X2Hxh

I collected this BT with loose_route() being called from my script, but not 
validate_dialog() or fix_route_dialog(): http://pastebin.com/6V7yPaHF

This BT was collected with all three functions being called from my script: 
http://pastebin.com/fZYYdndn


Ben Newlin

From: Bogdan-Andrei Iancu 
Date: Wednesday, July 27, 2016 at 3:57 AM
To: OpenSIPS users mailling list , "Newlin, Ben" 

Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Hi Ben,

First, if you use TH, makes no sense to do Record-Routing - there are 2 SIP 
concepts that overlaps. You either act as an end-point (by doing TH), either as 
a proxy (doing RR).

If doing TH, makes no sense to use validate + fix as these functions check and 
repair the routing information in the request (like Route and Contact headers). 
if you do TH, this routing info is actually hidden and added by OpenSIPS, so 
there is nothing to fix and repair.

Nevertheless, this should not crash or corrupt OpenSIPS. HAve you managed to 
get a corefile ?

Also if you suspect memory corruption, you can compile-in the memory debugger - 
see http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 23:20, Newlin, Ben wrote:
I have had 3 OpenSIPS server crashes in the last week. All were due to 
segmentation faults. I was not able to capture core dumps; I am configuring 
that now to catch the next crash.

My logs leading up to the crash are full of errors from fix_route_dialog() 
complaining about invalid URIs for sequential requests:

Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed:  (4) 
/ > (44)

Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed:  (4) 
/  (65)

Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed:  (4) 
/  (44)

Many times the “URI” displayed in the error message is actually internal 
OpenSIPS variables, as in the last error above. When they are from the SIP 
message, I have verified that the messages themselves are correctly formatted. 
This leads me to believe there is memory corruption occurring.

This all started when I updated my load-balancer servers to use Record-Routing, 
specifically the “double_rr” mechanism for when multiple interfaces exist. The 
Record-Routing is occurring on different servers which have not crashed. Only 
the servers 

Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?

[Johan De Clercq]

in the startup route, you should find  your pub ip using curl.
Put that in a var and then use that var in rtpproxy.

2016-07-27 15:07 GMT+02:00 Rodrigo Pimenta Carvalho :

> Dear OpenSIPS-users.
>
>
>
> The function nat_uac_test() (from module NATHELPER) works very well and
> tells me whether a cliente to my OpenSIPS is behind a NAT.
>
>
> But, for my specific network topology, is my OpenSIPS that is behind a
> NAT, from the client perspective. In this case I have to fix the SDP
> content that goes from OpenSIPS to the client, so that the client will be
> able to send its media to a public IP, when communicating to a peer in the
> same network domain of this SIP server.
>
>
> How can I be sure that for a client perspective the OpenSIPS is behind a
> NAT? In another words, is there a way to the OpenSIPS determine if its
> client is in the same network or in a remote network?
>
>
> --
>
>
> P.S. I suspect that OpenSIPS should be used in a node with public IP to
> simplify our solution. But our customer asked us to put OpenSIPS in a
> residential device that will be always behind a NAT for some smartphones
> perspective and not behind a NAT for another home devices perspective.
>
>
>
> ---
>
>
> Any hint will be very helpful!
>
>
> Best regards.
>
>
>
> 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
>
>
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Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

[Newlin, Ben]

Bogdan,

This is a different scenario than the other you responded to. As I said, we 
have two types of servers that work together. One is a load-balancer and runs 
as a proxy. It uses double Record-Route because it sends messages between 
public and private networks. Then we have our other servers using TH which 
receive those requests. We are not using TH and RR on the same server (although 
I would like to).

If validate_dialog() and fix_route_dialog() (and possibly loose_route()) should 
not be called when using TH, I believe the documentation should reference that. 
It states that match_dialog() must be used with TH, but does not indicate that 
the other functions should not be used or that the functionality won’t work. 
There is also no documentation of the incompatibility between RR and TH.

Either way, I ran a test where I removed all calls to loose_route(), 
validate_dialog(), and fix_route_dialog() from my script. The crash still 
occurred and the BT still pointed to fix_route_dialog() function. So it must be 
getting called from within Dialog module somewhere. That BT is here: 
http://pastebin.com/wu2X2Hxh

I collected this BT with loose_route() being called from my script, but not 
validate_dialog() or fix_route_dialog(): http://pastebin.com/6V7yPaHF

This BT was collected with all three functions being called from my script: 
http://pastebin.com/fZYYdndn


Ben Newlin

From: Bogdan-Andrei Iancu 
Date: Wednesday, July 27, 2016 at 3:57 AM
To: OpenSIPS users mailling list , "Newlin, Ben" 

Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Hi Ben,

First, if you use TH, makes no sense to do Record-Routing - there are 2 SIP 
concepts that overlaps. You either act as an end-point (by doing TH), either as 
a proxy (doing RR).

If doing TH, makes no sense to use validate + fix as these functions check and 
repair the routing information in the request (like Route and Contact headers). 
if you do TH, this routing info is actually hidden and added by OpenSIPS, so 
there is nothing to fix and repair.

Nevertheless, this should not crash or corrupt OpenSIPS. HAve you managed to 
get a corefile ?

Also if you suspect memory corruption, you can compile-in the memory debugger - 
see http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 23:20, Newlin, Ben wrote:
I have had 3 OpenSIPS server crashes in the last week. All were due to 
segmentation faults. I was not able to capture core dumps; I am configuring 
that now to catch the next crash.

My logs leading up to the crash are full of errors from fix_route_dialog() 
complaining about invalid URIs for sequential requests:

Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed:  (4) 
/ > (44)

Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed:  (4) 
/  (65)

Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed:  (4) 
/  (44)

Many times the “URI” displayed in the error message is actually internal 
OpenSIPS variables, as in the last error above. When they are from the SIP 
message, I have verified that the messages themselves are correctly formatted. 
This leads me to believe there is memory corruption occurring.

This all started when I updated my load-balancer servers to use Record-Routing, 
specifically the “double_rr” mechanism for when multiple interfaces exist. The 
Record-Routing is occurring on different servers which have not crashed. Only 
the servers receiving the Record-Routed messages are experiencing the errors.

Here is a piece of the code processing sequential requests. I am using the 
topology_hiding() functionality of the Dialog module. Are validate_dialog() and 
fix_route_dialog() still valid in a topology_hiding scenario?

if (t_check_trans())
setflag(SEQ_REQUEST);

  if (has_totag())
  {
loose_route();

if (match_dialog())
{
  if (!validate_dialog())
fix_route_dialog();

  if (is_method("BYE"))
setflag(ACC_FLAG);

  setflag(SEQ_REQUEST);
}
else if (!isflagset(SEQ_REQUEST))
{
  if (!is_method("ACK")) {
route(rlog, LV_ERROR, "check_sequential", "Sequential request not 
matched");
route(reply_error, "481", "Call Does Not Exist");
  }

  return(EXIT);
}
  }

I will attempt to get core dumps of future crashes.

Thanks,
Ben Newlin




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Re: [OpenSIPS-Users] Provisional response handling in case of forking

[Ramachandran, Agalya (Contractor)]

Hi Bogdan,

Yes you are correct. Am trying to implement that logic. (keep and log the first 
180, drop the others)
Am giving you snippet of onreply_route and relay2 attached here with this mail.

My intention is just to log the response sent by OpenSIPS. For other responses 
logic seems good and I could see that response sent out are logged.
Only for 180 resposne, am not meeting my condition.
Correct me if am wrong.

Regards,
Agalya
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Wednesday, July 27, 2016 3:22 AM
To: Ramachandran, Agalya (Contractor) ; 
OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking

Hi Agalya,

First, take care that you can set and trigger only one reply route per 
transaction !

Now, looking at your logic in the script, it does not reflect your statement : 
keep and log the first 180, drop the others <- is this correct ?

Best regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 21:26, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

I have a question related to the below topic.
When am receiving more than  180 response from different destinations , am 
sending 1st 180 to the caller and dropping consecutive  180 's.
And this logic is working fine.

Now I have a scenario, where I need to log what are the responses sent out of 
Opensips.
My logic for this is, am checking the flag value. If flag is set, log the 1st 
180 response, reset the flag and drop the remaining 180 response.

I could see FLAG_180 is set, and when I try to reset flag is not getting reset. 
Is it not possible to reset the flag values when relay is done?
I have added a route(relay2) in onreply_route  Below is the snippet of it.
Please guide me if am doing something wrong.

route[relay2] {
if ($rs==100)
   drop();
if ($rs==180) {
  if (isflagset(FLAG_180)){
xlog("INFO:opensips: Flag is set 
\n");
resetflag(FLAG_180);
  }
  else{
  drop();
 }
  #setflag(FLAG_180);
}
xlog("INFO:opensips: Sending $rs resposne out 
\n");
}


Regards,
Agalya
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Monday, June 13, 2016 6:31 AM
To: OpenSIPS users mailling list 
; Ramachandran, 
Agalya (Contractor) 

Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking

Hi Agalya,

Use the onreply route (be sure to onreply_avp_mode to be set to 1 - see 
http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) in 
combination with flags, to record when the an 180 reply was set.

Like:

if ($rs==180) {
if (isflagset(FLAG_180))
drop();
setflag(FALG_180);
}

The onreply_avp_mode 1 will ensure that the onreply route will not overlap for 
2 replies .

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips for our project and we are currently using opensips as 
proxy.
Am forking the incoming sip call, to two destinations. It Rings in both Dest A 
and Dest B, as a result I get two 180 Ringing response from A and B.
I want to filter only the first incoming 180 Ringing response and send to the 
actual caller. Is there a way to do this in opensips config file?

I have seen drop() function which drops the complete provisional response. But 
in my case I have to forward one 180 Ringing to the caller.
Can it be achieved by the changes in config file? Please guide me.

Regards,
Agalya






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onreply_route[handle_nat] {

$avp(traced_user) = "1";
sip_trace();
   
if ($rs==180) {
  if (isflagset(FLAG_180)){
xlog("INFO:opensips: onreply_route: Dropping consecutive 180 
resposne\n");
drop();
  }
setflag(FLAG_180);
}
route(relay2);
}

route[relay2] {

if ($rs==100)
   drop();
if ($rs==180) {
  if (isflagset(FLAG_180)){
xlog("INFO:opensips: Flag is set \n");
resetflag(FLAG_180);
  }
  else{
xlog("INFO:opensips: Flag is not set \n");
 drop();
  }
  #setflag(FLAG_180);
}
xlog("INFO:opensips: Sending $rs resposne out \n");
}
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[OpenSIPS-Users] name of event in the event message's body

[Чалков Артём]

Hi all!I trying to send some data by event_interface via rabbitmq. For do that, i use subscribe_event("EVENT_NAME", "rabbitmq:someaddress"), and then raise_event("EVENT_NAME", $var(eventdata)).All works fine, but the body of message, sent by raise_event, looks like "EVENT_NAME eventdata", but i need to send only eventdata, without EVENT_NAME. Is there some way to not send name of raised event in the body of message?

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[OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?

[Rodrigo Pimenta Carvalho]

Dear OpenSIPS-users.



The function nat_uac_test() (from module NATHELPER) works very well and tells 
me whether a cliente to my OpenSIPS is behind a NAT.


But, for my specific network topology, is my OpenSIPS that is behind a NAT, 
from the client perspective. In this case I have to fix the SDP content that 
goes from OpenSIPS to the client, so that the client will be able to send its 
media to a public IP, when communicating to a peer in the same network domain 
of this SIP server.


How can I be sure that for a client perspective the OpenSIPS is behind a NAT? 
In another words, is there a way to the OpenSIPS determine if its client is in 
the same network or in a remote network?


--


P.S. I suspect that OpenSIPS should be used in a node with public IP to 
simplify our solution. But our customer asked us to put OpenSIPS in a 
residential device that will be always behind a NAT for some smartphones 
perspective and not behind a NAT for another home devices perspective.


---


Any hint will be very helpful!


Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

[Newlin, Ben]

I understand that normally you would not need RR with TH, but the two concepts 
are not mutually exclusive in SIP. As I said, I have a need to Record-Route the 
call on my server as I am advertising a different address than I am listening 
on. This means that TH will populate the Contact header with the advertised 
address and if I cannot Record-Route with the actual address then I will not 
receive sequential requests.


Ben Newlin

From: Bogdan-Andrei Iancu 
Date: Wednesday, July 27, 2016 at 3:59 AM
To: OpenSIPS users mailling list , "Newlin, Ben" 

Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

As I mentioned in different thread, TH is not compatible with the RR mechanism. 
If you do TH, your OpenSIPS will act as and end point (from SIP perspective), 
so there will be no Route/RR headers at all. So no need to do loose_route or 
so. You just do TH matching for the sequential requests and nothing more.

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 22.07.2016 16:48, Newlin, Ben wrote:
Hi,

I am using the Dialog module with topology_hiding() in my server and I have a 
need to Record-Route the call on my server as I am advertising a different 
address than I am listening on. I have found what I believe is an inconsistency 
in the handling of Record-Route within the Dialog topology_hiding 
functionality. The topology_hiding isn’t a true B2BUA, but it does set up 
different parameters for the incoming UAC and outgoing UAS sides of the call 
for the Via headers, Record-Route and Route headers, and the Contact header(s).

The problem is that the record_route() and loose_route() functions operate on 
different sides of the call. The record_route() function will only add a 
Record-Route header to the outgoing UAS side of the call. And since the 
record_route() function cannot be called from onreply_route, but is no way to 
add a Record-Route header to the UAC side of the call.

On the other hand, the loose_route() function only operates on the incoming UAC 
side of the call and there is no way to perform loose_route() on the UAS side 
of the call.

So there is a situation where Record-Route headers can only be added on the 
outgoing UAS side, but the associated Route headers can only be removed on the 
incoming UAC side (where they won’t exist since they can’t be added) and any 
added headers on the UAS side cannot be processed properly due to the lack of 
loose_route.

I can provide further information if this is unclear. It should be easily 
reproducible by attempting to use record_route in a topology_hiding scenario. 
The route is added to the outbound leg, but is not removed by loose_route so 
the message is looped back every time.

Ben Newlin | Sr Voice Network Engineer, PureCloud
phone & fax +1.317.957.1009 | ben.new...@inin.com
[mage removed by sender.]
www.inin.com





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Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

[Bogdan-Andrei Iancu]

Hi Ben,

As I mentioned in different thread, TH is not compatible with the RR 
mechanism. If you do TH, your OpenSIPS will act as and end point (from 
SIP perspective), so there will be no Route/RR headers at all. So no 
need to do loose_route or so. You just do TH matching for the sequential 
requests and nothing more.


Regards,

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

On 22.07.2016 16:48, Newlin, Ben wrote:


Hi,

I am using the Dialog module with topology_hiding() in my server and I 
have a need to Record-Route the call on my server as I am advertising 
a different address than I am listening on. I have found what I 
believe is an inconsistency in the handling of Record-Route within the 
Dialog topology_hiding functionality. The topology_hiding isn’t a true 
B2BUA, but it does set up different parameters for the incoming UAC 
and outgoing UAS sides of the call for the Via headers, Record-Route 
and Route headers, and the Contact header(s).


The problem is that the record_route() and loose_route() functions 
operate on different sides of the call. The record_route() function 
will only add a Record-Route header to the outgoing UAS side of the 
call. And since the record_route() function cannot be called from 
onreply_route, but is no way to add a Record-Route header to the UAC 
side of the call.


On the other hand, the loose_route() function only operates on the 
incoming UAC side of the call and there is no way to perform 
loose_route() on the UAS side of the call.


So there is a situation where Record-Route headers can only be added 
on the outgoing UAS side, but the associated Route headers can only be 
removed on the incoming UAC side (where they won’t exist since they 
can’t be added) and any added headers on the UAS side cannot be 
processed properly due to the lack of loose_route.


I can provide further information if this is unclear. It should be 
easily reproducible by attempting to use record_route in a 
topology_hiding scenario. The route is added to the outbound leg, but 
is not removed by loose_route so the message is looped back every time.


*Ben Newlin***| Sr Voice Network Engineer, PureCloud

phone & fax +1.317.957.1009 | ben.new...@inin.com

www.inin.com 



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Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

[Bogdan-Andrei Iancu]

Hi Ben,

First, if you use TH, makes no sense to do Record-Routing - there are 2 
SIP concepts that overlaps. You either act as an end-point (by doing 
TH), either as a proxy (doing RR).


If doing TH, makes no sense to use validate + fix as these functions 
check and repair the routing information in the request (like Route and 
Contact headers). if you do TH, this routing info is actually hidden and 
added by OpenSIPS, so there is nothing to fix and repair.


Nevertheless, this should not crash or corrupt OpenSIPS. HAve you 
managed to get a corefile ?


Also if you suspect memory corruption, you can compile-in the memory 
debugger - see 
http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .


Regards,

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

On 26.07.2016 23:20, Newlin, Ben wrote:


I have had 3 OpenSIPS server crashes in the last week. All were due to 
segmentation faults. I was not able to capture core dumps; I am 
configuring that now to catch the next crash.


My logs leading up to the crash are full of errors from 
fix_route_dialog() complaining about invalid URIs for sequential requests:


Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse 
SIP uri


Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: 
 (4) / > (44)


Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse 
SIP uri


Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed: 
 (4) / 
 (65)


Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse 
SIP uri


Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: 
 (4) /  (44)


Many times the “URI” displayed in the error message is actually 
internal OpenSIPS variables, as in the last error above. When they are 
from the SIP message, I have verified that the messages themselves are 
correctly formatted. This leads me to believe there is memory 
corruption occurring.


This all started when I updated my load-balancer servers to use 
Record-Routing, specifically the “double_rr” mechanism for when 
multiple interfaces exist. The Record-Routing is occurring on 
different servers which have not crashed. Only the servers receiving 
the Record-Routed messages are experiencing the errors.


Here is a piece of the code processing sequential requests. I am using 
the topology_hiding() functionality of the Dialog module. Are 
validate_dialog() and fix_route_dialog() still valid in a 
topology_hiding scenario?


if (t_check_trans())

setflag(SEQ_REQUEST);

  if (has_totag())

  {

loose_route();

if (match_dialog())

{

  if (!validate_dialog())

fix_route_dialog();

  if (is_method("BYE"))

setflag(ACC_FLAG);

setflag(SEQ_REQUEST);

}

else if (!isflagset(SEQ_REQUEST))

{

  if (!is_method("ACK")) {

route(rlog, LV_ERROR, "check_sequential", "Sequential request not 
matched");


  route(reply_error, "481", "Call Does Not Exist");

  }

return(EXIT);

}

  }

I will attempt to get core dumps of future crashes.

Thanks,

Ben Newlin



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Re: [OpenSIPS-Users] How to tell if uac_auth() is successful

[Bogdan-Andrei Iancu]

Hi Ben

The uac_auth() should return true if the authentication was successfully 
performed. Are you sure you correctly perform the test on the return 
code? Do you see any error log from the function ?


Regards,

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

On 25.07.2016 23:00, Newlin, Ben wrote:


I am using the UAC and UAC_AUTH modules to perform trunk 
authentication. The uac_auth() function will only use credentials when 
the realm matches one in the 401/407 challenge. Is there any way to 
tell whether there was a successful authentication match?


The function doesn’t provide any return code (returns false even if 
match was found). I have tried checking for values in Authorization 
header ($ar, $auth.resp) but they are not populated.


If I can’t tell whether appropriate credentials were found, then I 
must always send another INVITE which might not have any Authorization 
and will just be rejected again.


Any help would be appreciated.

Thanks,

Ben Newlin



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Re: [OpenSIPS-Users] Save gateway used on ACC Table when using drouting

[Bogdan-Andrei Iancu]

Hi,

If the $avp(gw_id) is to hold the GW ID from DB table, you should not 
use it in do_routing() call - as you built that do_routing(), the 
$avp(gw_id) will be populated with the attribute field of the selected 
GW, overwriting the already set GW ID.


Regards,

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

On 25.07.2016 19:01, Annus Fictus wrote:

hello,

I tried with:

if (do_routing("$avp(gw_id)")) {
xlog("L_NOTICE","Gateway $avp(gw_id)");

But $avp(gw_id) is always empty.

On the module parameters:

modparam("drouting", "gw_id_avp", '$avp(gw_id)')

Any hint?

Regards


El 25/07/2016 a las 17:31, Jim DeVito escribió:

Sure thing!

http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id294247

Then just add to the acc module *_extra 
"carrier_id=$avp(carrier_id_avp)" carrier_id being the column in your 
acc database or log.


Thanks!!

---
Jim DeVito


On 2016-07-25 08:20, Annus Fictus wrote:

Hello,

I'd like know if is possible save in the acc table the used gateway
from drouting module to terminate a call.

Regards


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Re: [OpenSIPS-Users] Provisional response handling in case of forking

[Bogdan-Andrei Iancu]

Hi Agalya,

First, take care that you can set and trigger only one reply route per 
transaction!


Now, looking at your logic inthe script, it does not reflect your 
statement : keep and log the first 180, drop the others <- is this correct ?


Best regards,

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

On 26.07.2016 21:26, Ramachandran, Agalya (Contractor) wrote:


Hi Bogdan,

I have a question related to the below topic.

When am receiving more than  180 response from different destinations 
, am sending 1^st 180 to the caller and dropping consecutive  180 ‘s.


And this logic is working fine.

Now I have a scenario, where I need to log what are the responses sent 
out of Opensips.


My logic for this is, am checking the flag value. If flag is set, log 
the 1^st 180 response, reset the flag and drop the remaining 180 response.


I could see FLAG_180 is set, and when I try to reset flag *is not 
getting reset*. Is it not possible to reset the flag values when relay 
is done?


I have added a route(relay2) in onreply_route  Below is the snippet of 
it.


Please guide me if am doing something wrong.

route[relay2] {

if ($rs==100)

   drop();

if ($rs==180) {

  if (isflagset(FLAG_180)){

xlog("INFO:opensips: Flag is set \n");

resetflag(FLAG_180);

  }

  else{

 drop();

 }

#setflag(FLAG_180);

}

xlog("INFO:opensips: Sending $rs resposne out \n");

}

Regards,
Agalya

*From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
*Sent:* Monday, June 13, 2016 6:31 AM
*To:* OpenSIPS users mailling list ; 
Ramachandran, Agalya (Contractor) 
*Subject:* Re: [OpenSIPS-Users] Provisional response handling in case 
of forking


Hi Agalya,

Use the onreply route (be sure to onreply_avp_mode to be set to 1 - 
see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) 
in combination with flags, to record when the an 180 reply was set.


Like:

if ($rs==180) {
if (isflagset(FLAG_180))
drop();
setflag(FALG_180);
}

The onreply_avp_mode 1 will ensure that the onreply route will not 
overlap for 2 replies .


Regards,

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

On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote:

Hi team,

We are using opensips for our project and we are currently using
opensips as proxy.

Am forking the incoming sip call, to two destinations. It Rings in
both Dest A and Dest B, as a result I get two 180 Ringing response
from A and B.

I want to filter only the first incoming 180 Ringing response and
send to the actual caller. Is there a way to do this in opensips
config file?

I have seen drop() function which drops the complete provisional
response. But in my case I have to forward one 180 Ringing to the
caller.

Can it be achieved by the changes in config file? Please guide me.

Regards,
Agalya




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