Re: [OpenSIPS-Users] Add more bodies to INTITE

[Răzvan Crainea]

Hi, Denis!

1. Yes
2. Yes, the function adds the separator.

Best regards,

Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com

On 12/06/2017 06:32 AM, Denis via Users wrote:

Hello, Razvan!
Thank you.
And yet two things:
1) Can i use rest_client Module for getting prepared xml document for 
my task? Will this work?
2) In the example i can see boundary entity. As i understand this 
entity serving to divide several bodies of the message. Will 
"add_body_part()" insert it automatically or i should do it manually?

Thank you.
--
С уважением, Денис.
Best regards, Denis
05.12.2017, 18:37, "Răzvan Crainea" :

Hi, Denis!

Check the new xml module[1]. You can use it to build the XML in the 
example.


[1] http://www.opensips.org/html/docs/modules/2.3.x/xml.html

Best regards,
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com 
On 12/05/2017 04:36 PM, Denis via Users wrote:

Hello, Razvan!
I want to ask to analyze the example. Can i add exactly such body, 
in the such format, and if possible, how?

In [1] we have only one param. "/body/". How can i from this param?
Thank you.
--
С уважением, Денис.
Best regards, Denis
05.12.2017, 17:21, "Răzvan Crainea"  
:
Yes, starting with OpenSIPS 2.3 you can add a body part using the 
add_body_part() function[1].


[1] 
http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops.html#idp5587328


Best regards,
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com 
On 12/05/2017 11:02 AM, Denis via Users wrote:

Hello!
Opensips 2.3
In the link below you can find an example of the INVITE which i 
want to send to some callee. Before, i received the INVITE from 
caller with only SDP body. I.e. i need to add some additional body 
to the INVITE with structure as shown in the example.

https://yadi.sk/i/389OOlLX3QKx27
Can i do it, using Opensips?
Thank you for any help.
--
С уважением, Денис.
Best regards, Denis
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[OpenSIPS-Users] Is OpenSIPS fast to stop calling tone in forked calls?

[Rodrigo Pimenta Carvalho]

Hi.


When my OpenSIPS has just 2 peers on-line (user A and user B ), if A calls B, A 
will listen the calling tone just until B answers. So, if B sends a SIP OK, the 
calling tone will be immediately stopped.


When user B is on-line in several devices (more than 1 contact for the same 
AOR), let's say 2 mobile phones and 2 desktops, 4 devices will ring and the 
user A will listen the calling tone normally. But, when B answers (in any 
device), there will be some calling tones still to be played in the A's device. 
That is, when the number of called devices increases for a same called 
subscriber,  it seems that OpenSIPS become slower to stop the calling tone. But 
it is fast enough to stop the ring tones in the others devices.


Is it a matter of OpenSIPS configurations file? (opensips.cfg)

If yes, could someone point me what part of my configuration should I change or 
review?


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 @ ClueCon Weekly

[Bogdan-Andrei Iancu]

Heads up, one our left to the ClueCon Weekly with OpenSIPS .

To join us, dial 888 at https://conference.freeswitch.org/vc/ 



Regards,

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

On 12/04/2017 07:38 PM, Răzvan Crainea wrote:
I am glad to announce you that Bogdan Iancu will be talking this week 
at the FreeSWTICH's ClueCon Weekly conference call[1]. If you want to 
find out the latest news about OpenSIPS, try to join this awesome conf 
call!
The event is scheduled for this Wednesday, 6th of December 2017, at 
12:00 Central Time[1]. The video call will also be streamed live on 
YouTube[2].


[1] https://www.facebook.com/events/127055664628500/
[2] https://www.youtube.com/watch?v=XtMh9rAtB8w&feature=youtu.be

See you on Wednesday!



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