Ok I got it - endpoint must send MESSAGE to sip:conference-name@domain.name,
but initially it sends to conference-name.
So as I understood - if (is_method("MESSAGE") &&
!(starts_with("$fU",
"chat"))) should make everything work correct.
Sorry for emotions :)
2015-12-15 19:00 GMT+02:00 Alexandru Covalschi <568691(a)gmail.com>om>:
upd: Let me describe my use case. I need conference
chats. Clients are
1001-1009(a)domain.name
Conference is 3500(a)domain.name
Messages are sent from 1001-1009(a)domain.name to 3500(a)domain.name
Every user joins conference room chat-3500. Join is successful - but
nothing more. Messages are not relayed anywhere and I don't want to relay
them - I just need to understand how this module works. My current
configuration is:
if (is_method("MESSAGE") && $fU != "chat-3500")
{
if(imc_manager())
sl_send_reply("200", "ok");
else
sl_send_reply("500", "command error");
exit;
}
This allows me to receive system messages - but I can't get any messages
from clients.
2015-12-15 18:43 GMT+02:00 Alexandru Covalschi <568691(a)gmail.com>om>:
Hello again
First of all I wanted to ask if someone ever implemented that
http://kamailio.org/docs/modules/4.3.x/modules/imc.html with WebRTC
Second question is - I don't understand the logic. In description is
said:
Handles Message method.It detects if the body of the message is a
conference command.If so it executes it, otherwise it sends the message to
all the members in the room.
But why in example (well however it has broken syntax) to IMC manager are
sent only messages from chat-rooms? How message from client can possibly
reach imc-manager then?
Also - when I send message with body "1111111" from user 1001 and
imc_manager catches it - I receive 500 command error. Why? :/
All that is working on top of sipjs-demo.
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web:
http://abs-telecom.com/
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web:
http://abs-telecom.com/
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web:
http://abs-telecom.com/