[jira] [Updated] (OPENMEETINGS-1860) SIP Integration - Can't enter conference room (leader wait)

["Maxim Solodovnik (JIRA)" <ji...@apache.org>]

     [ https://issues.apache.org/jira/browse/OPENMEETINGS-1860?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Maxim Solodovnik updated OPENMEETINGS-1860:
-------------------------------------------
    Fix Version/s:     (was: 4.0.5)
                   4.0.6

> 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
>             Fix For: 4.0.6
>
>         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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