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
