DB credentials used are configured in /etc/odbc.ini
as described in http://openmeetings.apache.org/red5sip-integration_3.0.html
On 10 August 2014 22:09, Horace Miles <horace.mi...@myit-solutions.com>
wrote:
> I took the firewall down. Still had the same problem.
>
>
>
> I reverse the logic in this line of the extensions.conf
>
> exten =>
> _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?notavail:ok)
> >>>>>>>>>> reverse the notavail and OK.
>
> It will then invoke the video bridge. Cant make a call but it shows me
> that it asterisk is not finding the record in openmeetings. I cant figure
> out wht account it is using to make the query. Openmeetings, root, or
> red5sip_user. I cant figure out the correlation of which is making the
> call to the openmeetings database.
>
>
>
> Miles
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Sunday, August 10, 2014 3:36 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Maybe it is network configuration issue as described here:
> http://stackoverflow.com/questions/22093328/asterisk-sip-retransmission-timeout
> ?
>
>
>
> can you check it with all firewalls disabled?
>
>
>
> On 8 August 2014 23:51, Horace Miles <horace.mi...@myit-solutions.com>
> wrote:
>
> Maxim,
>
>
>
> Whenever you have time I understand. Here are all of my configurations by
> file name. I hope it will help.
>
> HERE IS THE CONFIGURATION FOR EACH FILE FOR THE ASTERISK INTEGRATION. I
> HAVE SEPPERATED EACH SECTION WITH “==========”
>
> HOPEFULLY SOMEONE CAN SEE THE ERROR I HAVE MADE
>
>
>
> Miles
>
> ================================================================
>
> CONFIGURATION for /etc/odbc.ini
>
> [asterisk-connector]
>
> Description = MySQL connection to 'openmeetings' database
>
> Driver = MySQL
>
> Database = open30
>
> Server = 127.0.0.1
>
> USER = root
>
> PASSWORD =******
>
> Port = 3306
>
> Socket = /var/run/mysqld/mysqld.sock
>
> ================================================================
>
> CONFIGURATION for /etc/odbcinst.ini
>
> [MySQL]
>
> Description = ODBC for MySQL
>
> Driver = /usr/lib/i386-linux-gnu/odbc/libmyodbc.so
>
> Setup = /usr/lib/i386-linux-gnu/odbc/libodbcmyS.so
>
> FileUsage = 1
>
> ================================================================
>
> CONFIGURATION for in /etc/asterisk/modules.conf
>
> [modules]
>
> autoload=yes
>
> ;
>
> ; Any modules that need to be loaded before the Asterisk core has been
>
> ; initialized (just after the logger has been initialized) can be loaded
>
> ; using 'preload'. This will frequently be needed if you wish to map all
>
> ; module configuration files into Realtime storage, since the Realtime
>
> ; driver will need to be loaded before the modules using those
> configuration
>
> ; files are initialized.
>
> ;
>
> ; An example of loading ODBC support would be:
>
> preload => res_odbc.so
>
> preload => res_config_odbc.so
>
> ================================================================
>
> CONFIGURATION for /etc/asterisk/res_odbc.conf
>
> ;;; odbc setup file
>
>
>
> ; ENV is a global set of environmental variables that will get set.
>
> ; Note that all environmental variables can be seen by all connections,
>
> ; so you can't have different values for different connections.
>
> [ENV]
>
> ;INFORMIXSERVER => my_special_database
>
> ;INFORMIXDIR => /opt/informix
>
> ;ORACLE_HOME => /home/oracle
>
>
>
> ; All other sections are arbitrary names for database connections.
>
>
>
> ;
>
> ; The context name is what will be used in other configuration files, such
>
> ; as extconfig.conf and func_odbc.conf, to reference this connection.
>
> [asterisk]
>
> ;
>
> ; Permit disabling sections without needing to comment them out.
>
> ; If not specified, it is assumed the section is enabled.
>
> enabled => yes
>
> ;
>
> ; This value should match an entry in /etc/odbc.ini
>
> ; (or /usr/local/etc/odbc.ini, on FreeBSD and similar systems).
>
> dsn => asterisk-connector
>
> ;
>
> ; Username for connecting to the database. The user defaults to the
> context
>
> ; name if unspecified.
>
> username => admin
>
> ;
>
> ; Password for authenticating the user to the database. The default
>
> ; password is blank.
>
> password => ******
>
> ;
>
> ; Build a connection at startup?
>
> pre-connect => yes
>
> ================================================================
>
> Configuration for /etc/asterisk/sip.conf
>
> ;
>
> ;
>
> ; SIP Configuration example for Asterisk
>
> ;
>
> ; Note: Please read the security documentation for Asterisk in order to
>
> ; understand the risks of installing Asterisk with the sample
>
> ; configuration. If your Asterisk is installed on a public
>
> ; IP address connected to the Internet, you will want to learn
>
> ; about the various security settings BEFORE you start
>
> ; Asterisk.
>
> ;
>
> ; Especially note the following settings:
>
> ; - allowguest (default enabled)
>
> ; - permit/deny/acl - IP address filters
>
> ; - contactpermit/contactdeny/contactacl - IP
> address filters for registrations
>
> ; - context - Which set of services you offer
> various users
>
> ;
>
>
>
> [general]
>
> context=public ; Default context for incoming calls.
> Defaults to 'default'
>
> allowoverlap=no ; Disable overlap dialing support.
> (Default is yes)
>
> realm=asterisk ; Realm for digest authentication
>
> udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to
> (0.0.0.0 binds to all)
>
> ; Optionally add a port number,
> 192.168.1.1:5062 (default is port 5060)
>
>
>
> tcpenable=yes ; Enable server for incoming TCP
> connections (default is no)
>
> tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to
> (0.0.0.0 binds to all interfaces)
>
> transport=udp ; Set the default transports. The order
> determines the primary default transport.
>
> ; If tcpenable=no and the transport set is
> tcp, we will fallback to UDP.
>
>
>
> srvlookup=yes ; Enable DNS SRV lookups on outbound calls
>
> maxexpiry=43200 ; Maximum allowed time of incoming
> registrations (seconds)
>
> videosupport=yes ; Turn on support for SIP video. You need
> to turn this
>
> rtcachefriends=yes ; Cache realtime friends by adding them to
> the internal list
>
>
>
> ;domain=mydomain.tld,mydomain-incoming
>
> ; Add domain and configure incoming context
>
> ; for external calls to this domain
>
> domain=127.0.0.1 ; Add IP address as local domain
>
> domain=98.174.244.232 ; You can have several "domain" settings
>
>
>
> [basic-options](!) ; a template
>
> dtmfmode=rfc2833
>
> context=from-office
>
> type=friend
>
>
>
> [natted-phone](!,basic-options) ; another template inheriting
> basic-options
>
> directmedia=no
>
> host=dynamic
>
>
>
> [public-phone](!,basic-options) ; another template inheriting
> basic-options
>
> directmedia=yes
>
>
>
> [my-codecs](!) ; a template for my preferred codecs
>
> disallow=all
>
> allow=ilbc
>
> allow=g729
>
> allow=gsm
>
> allow=g723
>
> allow=ulaw
>
> ; Or, more simply:
>
> ;allow=!all,ilbc,g729,gsm,g723,ulaw
>
>
>
> [ulaw-phone](!) ; and another one for ulaw-only
>
> disallow=all
>
> allow=ulaw
>
> ; Again, more simply:
>
> ;allow=!all,ulaw
>
>
>
> ; and finally instantiate a few phones
>
> ;
>
> ; [2133](natted-phone,my-codecs)
>
> ; secret = peekaboo
>
> ; [2134](natted-phone,ulaw-phone)
>
> ; secret = not_very_secret
>
> ; [2136](public-phone,ulaw-phone)
>
> ; secret = not_very_secret_either
>
> ; ...
>
> ;
>
> [red5sip_user]
>
> type=friend
>
> secret=12345
>
> disallow=all
>
> allow=ulaw
>
> allow=h264
>
> host=dynamic
>
> nat=no
>
> ;nat=force_rport,comedia
>
> context=rooms-red5sip
>
> ================================================================
>
> CONFIGURATION FOR /etc/asterisk/extconfig.conf
>
> ;
>
> ; Static and realtime external configuration
>
> ; engine configuration
>
> ;
>
> ; See
> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
>
> ; for basic table formatting information.
>
> ;
>
> [settings]
>
> sippeers => odbc,asterisk,sipusers
>
> ================================================================
>
> CONFIGURATION FOR /etc/asterisk/extensions.conf
>
> [rooms]
>
> exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
>
> exten => _400X!,n(ok),SET(PIN=${DB(open30/room/${EXTEN})})
>
> exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
>
> exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
>
> exten => _400X!,n,Hangup
>
> exten => _400X!,n(notavail),Answer()
>
> exten => _400X!,n,Playback(invalid)
>
> exten => _400X!,n,Hangup
>
>
>
> [rooms-originate]
>
> exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
>
> exten => _400X!,n,Hangup
>
>
>
> [rooms-out]
>
> ; *****************************************************
>
> ; Extensions for outgoing calls from Openmeetings room.
>
> ; *****************************************************
>
>
>
> [rooms-red5sip]
>
> exten => _400X!,1,GotoIf($[${DB_EXISTS(open30/room/${EXTEN})}]?ok:notavail)
>
> exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
>
> exten => _400X!,n(notavail),Hangup
>
> ================================================================
>
> CONFIGURATION for /etc/asterisk/confbridge.conf
>
> [red5sip_user]
>
> type=user
>
> marked=yes
>
> dsp_drop_silence=yes
>
> denoise=true
>
>
>
> [sip_user]
>
> type=user
>
> end_marked=yes
>
> wait_marked=yes
>
> music_on_hold_when_empty=yes
>
> dsp_drop_silence=yes
>
> denoise=true
>
>
>
> [default_bridge]
>
> type=bridge
>
> video_mode=follow_talker
>
> ;video_mode=last_marked
>
> ;video_mode=first_marked
>
> ================================================================
>
> CONFIGURATION /etc/asterisk/manager.conf
>
> [general]
>
> ;enabled = no
>
> ;webenabled = yes
>
> enabled = yes
>
> webenabled = no
>
> port = 5038
>
> bindaddr = 127.0.0.1
>
>
>
> [openmeetings]
>
> secret = 12345
>
> deny=0.0.0.0/0.0.0.0
>
> permit=127.0.0.1/255.255.255.0
>
> read = all
>
> write = all
>
> ================================================================
>
> CONFIGURATION for
> /usr/lib/red53/webapps/konnectme/WEB-INF/classes/openmeetings-applicationContext.xml
>
>
>
> class="org.apache.openmeetings.db.dao.calendar.AppointmentCategoryDao" />
>
> <bean id="roommanagement"
> class="org.apache.openmeetings.data.conference.RoomManager" />
>
> <bean id="roomDao"
> class="org.apache.openmeetings.db.dao.room.RoomDao"/>
>
> <bean id="sipDao"
> class="org.apache.openmeetings.db.dao.room.SipDao">
>
> <!-- Should be uncommented and updated with real values for
> Asterisk -->
>
>
> <constructor-arg><value>127.0.0.1</value></constructor-arg>
>
>
> <constructor-arg><value>5038</value></constructor-arg>
>
>
> <constructor-arg><value>openmeetings</value></constructor-arg>
>
>
> <constructor-arg><value>12345</value></constructor-arg>
>
> ================================================================
>
> CONFIGURATION for /opt/red5sip/red5sip_3.0/settings.properties
>
> red5.host=127.0.0.1
>
> om.context=konnectme
>
> red5.codec=asao
>
> red5.codec.rate=22
>
> sip.obproxy=127.0.0.1
>
> sip.phone=red5sip_user
>
> sip.authid=red5sip_user
>
> sip.secret=12345
>
> sip.realm=asterisk
>
> sip.proxy=127.0.0.1
>
> rooms.forceStart=no
>
> rooms=1
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Wednesday, August 06, 2014 10:46 PM
>
>
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Hello Horace,
>
>
>
> sorry for keeping silence, a little bit bit busy right now
>
> SIP transport set up the bridge from asterisk to red5 and performs
> audio/video transcoding rtp <->rtmp
>
>
>
> according to your issue it seems like creadentials specified in settings
> file are invalid for your Asterisk, can it be a problem?
>
> Will try to reproduce your problem as soon as i will get some time
>
>
>
> On 7 August 2014 02:53, Horace Miles <horace.mi...@myit-solutions.com>
> wrote:
>
> Maxim,
>
> Perhaps if I knew exactly what sip transport does, I might be able to
> figure this out. Can you tell me what it is suppose to do..
>
>
>
>
>
> Miles
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Friday, August 01, 2014 8:22 PM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Simple test if everything works is:
>
> 1) go to Admin->Conference rooms
>
> 2) select room
>
> 3) Check enable SIP
>
> 4) SIP number should appear in room panel (maybe after save)
>
>
>
> is it works for you?
>
>
>
> On 2 August 2014 00:36, Horace Miles <horace.mi...@myit-solutions.com>
> wrote:
>
> Ok found red5sip.enable value = yes
>
> Asterisk is configured to access openmeeting database through
> asterisk-connector
>
> Bean as been uncommented in openmeetings-application.xml and configure
> with matching values in asterisk/manager.conf
>
> I have re-saved all users in Openmeetings to recreate password hashes in
> asterisk
>
> Sip is enabled in rooms that have been created.
>
>
>
> I can telnet to localhost 5080 and 1935
>
>
>
> I am still having the following problems
>
> Sip Transport will not stay in the room pops in and out every two seconds
>
> It appears as though the sip transport can register but is unable to
> receive the invite message.
>
> In the extension.conf
>
> I get the following
>
> n -- Executing [40016@rooms-red5sip:1]
> GotoIf(“SIP/red5sip_user-000000a6”,”0?ok:notavail”) in new stack
>
> n -- Goto (rooms-red5sip,40016,3)
>
> n --Executing [40016@rooms-red5sip:3]
> Hangup(“/red5sip_user-000000a6”,””) in new stack
>
> n Spawn extension (rooms-red5sip, 40016,3) exited non-zero on
> ‘/red5sip_user-000000a6’
>
> It appears to check the database not find the room and then hang up.
>
> *From:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Friday, August 01, 2014 10:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> you can search red5sip in config :)
>
> the key is "red5sip.enable"
>
>
>
> On 1 August 2014 23:48, Horace Miles <horace.mi...@myit-solutions.com>
> wrote:
>
> Maxim thanks for the response.
>
> I have confirmed everything but I am not sure where to find this setting.
> I am assuming Rootconfig is Openmeeting Admin->Configuration. If so I
> don’t a setting for Red5sip key.
>
> 3) red5sip* key should be enabled in Admin->Config – NOT SURE OF THIS STEP
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Wednesday, July 30, 2014 6:07 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> OM is accessible on all network interfaces by default
>
> config.xml need to be modified only in case you need to restrict OM client.
>
>
>
> According to red5sip enter-exit-enter-exit-.... it should be due to
> misconfiguration. Unfortunately this integration is not simple by design :(
> I'm using logs and debug to set it up properly.
>
>
>
> Main steps are
>
> 1) asterisk should be configured to have access to OM DB
>
> 2) asterisk bean should be uncommented and configured properly in
> openmeetings-application.xml
>
> 3) red5sip* key should be enabled in Admin->Config
>
> 4) in case asterisk is integrated with OM user should be re-saved (to have
> password-hash being saved in asterisk DB table)
>
> 5) sip should be enabled in the room
>
>
>
> this should be all (hope I haven't miss anything)
>
>
>
> On 29 July 2014 08:29, Horace Miles <horace.mi...@myit-solutions.com>
> wrote:
>
> Hi Maxim,
>
> My box is connected directly to a public IP, no NAT. My understanding
> was that Openmeetings to be access from the internet needed to be on a
> public address. That address would be the one in the config.xml. If I a
> mistaken let me know.
>
> Can I have your thoughts on the following:
>
>
>
> I am unable to get the sip agent to bind to 127.0.0.1. It refuses to bind
> unless I have bind it to the same address that is in red5home
> /webapps/openmeetings/public/config.xml
>
>
>
> The problem appears to be either that the SIP protocol wants to use
> 127.0.0.1 for the subscribe or invites and SIP agent is bound to the Public
> IP address. Therefore it is generating the error for seqno 2 which would
> be the SIP Invite (I am assuming). I have not been able to get the SIP
> tansport to bind to 127.0.0.1 which would probably solve this problem.
>
>
>
> Your thoughts/
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Friday, July 25, 2014 7:22 AM
> *To:* Horace Miles
> *Subject:* Re: VOIP and Sip Integration
>
>
>
> hope you will be able to fix it, please let ne know if additional help is
> required
>
>
>
> On 25 July 2014 20:53, Horace Miles <horace.mi...@myit-solutions.com>
> wrote:
>
> Hey thanks for the files.
>
>
>
> I compared and I have found the following:
>
>
>
> It appears the integration is setup for for a box that is NAT’ed. I
> thought openmeetings had to be on a static public IP address?
>
>
>
> So I changed every place that is referencing 127.0.0.1 to my IP address.
>
>
>
> The Sip Agent/Openmeetings Manager does not come into the room until I
> restart Asterisk. I can see it successfully logging on and then
> immediately logging off. The room is successfully spawned.
>
>
>
> There seem to be a problem with the manager once it signs on with the sip
> handshake (again I am guessing)
>
>
>
> chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on
> transmission #########@127.0.0.1 for seqno 2 (Critical Response) see……
> Packet timed out afer 32000ms with no response.
>
>
>
> I will load wireshark later today on the PBX to see what else I might find.
>
>
>
> Thanks for all your help.
>
>
>
>
>
> *From:* Maxim Solodovnik [mailto:solomax...@gmail.com]
> *Sent:* Thursday, July 24, 2014 2:42 AM
> *To:* Openmeetings user-list
> *Subject:* Re: Pointer on WB
>
>
>
> Only with code modification
>
> On Jul 24, 2014 4:40 PM, "Raju M K" <mkraju...@gmail.com> wrote:
>
> Dear all,
>
> can i disable arrow pointer for all participants in restricted room on
> Whiteboard??
>
>
> --
> Regards,
> M K Raju.
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
