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:[email protected]]
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 <[email protected]> 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:[email protected]]
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 <[email protected]> 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:[email protected]]
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 <[email protected]> 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:[email protected]]
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 <[email protected]> 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:[email protected]]
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 <[email protected]> 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:[email protected]]
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 <[email protected]> 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 <mailto:%23#%23%23%23%23%23%23%[email protected]> 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:[email protected]]
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" <[email protected]> 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
