[Freeswitch-users] How to send SIP signaling message from fs console.

Michael Jerris mike at jerris.com
Fri May 22 19:52:27 MSD 2015


> On May 22, 2015, at 12:55 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> Michael, I think over how to organize the control of the call recording on OpenSips from FS dialplan (Scheme where FS is managing callcontrol, OpenSIPs RTP media processing)
> As an option I think to make it through
> 1) additional headers in the messages 100, 180, 183, 200 (start recording). Stop recording through RE-INVITE;
> 2) as a separate SIP INFO message (start, stop recording - more elegant solution).
> 
> Tell me, please, as much as possible
> 1) send a custom SIP INFO message 

from api interface: uuid_send_info_function
from dialplan: send_info

> 2) how custom headers can be added to 100, 180, 183, 200 messages

https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack#SofiaSIPStack-ChannelVariables <https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack#SofiaSIPStack-ChannelVariables>

> 
> On Thu, May 21, 2015 at 11:23 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Or its totally possible that I have asked for specifics multiple times now and you have ignored me every single time.  YOU CAN DO THIS IN FREESWITCH BUT I CAN NOT TELL YOU HOW UNLESS YOU TELL ME WHAT SIP MESSAGES YOU ARE TRYING TO SEND IN WHAT SITUATIONS!!!!!!!!
> 
> 
>> On May 21, 2015, at 4:14 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> 
>> Giovanni, 
>> 
>> Thank you very much, my intention is to know the way  to handle/manipulate (low level) sip packets with freeswitch if it possible. As per your reply this is obvious way to use Kamailio or OpenSIPS infront of FS. 
>> 
>> On Thu, May 21, 2015 at 4:31 PM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>> Naveen,
>> 
>> Please take note that FreeSWITCH is not intended to let the user to manipulate in arbitrary way SIP dialogs.
>> 
>> You can do that for specific purposes and in.specific cases, within specific boundaries.
>> 
>> If you are looking to interact with SIP directly and freely, you may want to look at Kamailio and OpenSIPS.
>> 
>> You can put one of them in front of FreeSWITCH, and you can cross command them, for example via.lua scripting (both of them and FreeSWITCH can be scripted in.lua).
>> 
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> 
>> On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> I am aware of uuid_hangup. Indeed my intention to know the way to send (low level)sip messages 
>> from fs console for a selected user. 
>> 
>> 
>> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
>>  I would like to reject the call when it ringing from the fs console. 
>> 
>> uuid_hangup
>>  
>> 
>> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> I am trying to do the following,  I would like to reject the call when it ringing from the fs console. 
>> And second thing is I am pretty much eager to know the way to send sip(signaling) message manually for a 
>> selected channel from fs console. 
>> 
>> 
>> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> respond is one way, what exact message are you trying to send, and at what point in the call.  There are capabilites to trigger re-invites in some situations, transfer, send notify or info or message.  It depends on what exactly you are trying to do.
>> 
>> 
>>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>>> 
>>> Hi, 
>>> 
>>> I am wondering how to send sip signaling message from the fs console for the particular user/caller. 
>>> I found  respond dialplan application to send sip message back to the caller. 
>>> Like 
>>> <action application="respond" data="480 Try again later"/>
>>> ​Is there any way to send sip message back to the caller. One use case is call rejection or playing with SIP​
>>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150522/c5ebfa55/attachment-0001.html 


Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users mailing list