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https://issues.apache.org/jira/browse/OPENMEETINGS-1860?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=16420279#comment-16420279
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Leonid commented on OPENMEETINGS-1860:
--------------------------------------
UPDATE:
OK, I've looked deeply in Asterisk's confbridge.conf:
[red5sip_user]; This is OM's user account used to connect rooms to Asterisk
(when SIP is enabled for this room), which is use to be *Leader* in a conference
type=user
marked=yes; Is this user should be marked by Asterisk as a leader in a
conference?
dsp_drop_silence=yes
denoise=true
[sip_user]; This is external SIP user (i.e. a trunk from another Asterisk
server) which is trying to get inside OM right into OM's conference room.
type=user
end_marked=yes
wait_marked=yes;Should the external SIP user wait for Leader to join the
conference
music_on_hold_when_empty=yes
dsp_drop_silence=yes
denoise=true
> SIP Integration - Can't enter conference room (leader wait)
> -----------------------------------------------------------
>
> Key: OPENMEETINGS-1860
> URL: https://issues.apache.org/jira/browse/OPENMEETINGS-1860
> Project: Openmeetings
> Issue Type: Bug
> Components: Audio/Video
> Affects Versions: 4.0.2
> Reporter: Leonid
> Assignee: Maxim Solodovnik
> Priority: Major
> Attachments: applicationContext.xml, confbridge.conf, extconfig.conf,
> extensions.conf, manager.conf, modules.conf, res_config_mysql.conf,
> settings.properties, sip.conf
>
>
> I've followed the official documentation and got a partial success on that.
> # Installation of OM 4.0.2 on Ubuntu 14.04 by Alvares Bustos [from this
> page|https://cwiki.apache.org/confluence/display/OPENMEETINGS/Tutorials+for+installing+OpenMeetings+and+Tools].
> *(Status: OK - Web Conferences went fine)*
> # [VoIP and SIP Integration for OM
> 4.0.2|https://openmeetings.apache.org/voip-sip-integration.html]. *(Status:
> OK - OM manager account can connect to Asterisk)*
> # Enabled red5sip in OM (*red5sip.enable == yes*)
> # I've created a room with SIP enabled, got the SIP-extension room number
> generated. *(Status: OK)*
> # Also, I've created additional extension on that Asterisk server.
> *(Status: OK - can register on the server with software SIP-Phone (Linphone,
> MicroSIP))*
> When I'm calling from SIP-phone extension (*#40099*) to OM-room extension
> number (*#40017*) I'm getting a message "*The conference will begin when the
> leader arrives*" and then I'm placing on *Hold* by Asterisk (I hear a
> music-on-hold).
> By the time I was trying to call via Asterisk there were several users
> (Admin, Moderate-rights user, User) right in the conference room. Also
> changed the room types - conference/presentation; private/public. Turned on
> PIN setting - I can enter the pin but then I get the same *Leader wait*
> message. Also tried exit/enter to conference room while I was on hold on SIP
> - no luck.
> Tried to change *rooms.forceStart* from *no* to *yes;* set up room ID in
> *rooms* option in *settings.properties* file of *red5sip* - no luck aslo.
> P.S. I did red5, red5sip, asterisk services restart every time I made changes
> in configs.
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