"When I configure config.xml as you have yours, I am not able to connect to the server. I get the connection time out error." this usually mean you have port 1935 closed closed on specific interface i.e. "telnet localhost 1935" will fail try to switch off your firewall and try again
On 30 July 2014 20:33, Horace Miles <horace.mi...@myit-solutions.com> wrote: > Maxim, > > > > When I configure config.xml as you have yours, I am not able to connect > to the server. I get the connection time out error. > > So my config.xml has my public IP address in it for rtmp and http settings. > > > > When I configure red5sip/ openmeetings properties settings > > Red5.host = ip address of rtmp and http settings in config.xml > > Sip.obproxy = ip address of rtmp and http settings in config.xml > > Sip.proxy= ip address of rtmp and http settings in config.xml > > > > Any other setting of 0.0.0.0 or 127.0.0.1 Red5sip fails to get the session > > > > The in Asterisk the openmeetings manager logs on and then right back up > which seems to coincide with the sip transport popping in and out of the > room. > > It appears that invite request done by the sip.transport is being > refused. Which is bound to 127.0.0.1. (I am so confused as to what is > going on.) > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Sunday, July 27, 2014 12:07 AM > *To:* Horace Miles; Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > Additionally red5sip connects to red5 server directly, not to the swf > client, so contents of config.xml is ignored while connecting by red5sip > > > > On 27 July 2014 12:37, Maxim Solodovnik <solomax...@gmail.com> wrote: > > Hello Horas, > > > > Please write to user mailing list. > > > > I currently have no host configured in > /webapps/openmeetings/public/config.xml file, all hosts are allowed > > Why do you need to limit host in this file? > > > > On 27 July 2014 04:44, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Hi Maxim, > > > > 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 8:44 AM > *To:* Horace Miles > *Subject:* Re: VOIP and Sip Integration > > > > uploaded > > > > On 24 July 2014 20:40, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Maxim, > > > > Thanks I appreciate it very much. > > > > I have created you an account on my cloud server : > http://mycloud.myit-solutions.com > > Login: mmaxim > > Password: chief123 > > > > There is a shared folder labeled openmeetings. You can upload the files > there. You have 5 GB of space. > > > > Let me know if you have any problems with this. I won’t be available > again until tonight but I will look at that time. > > > > Thanks a million > > > > Miles > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Wednesday, July 23, 2014 7:10 AM > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > I can privately send you all mine asterisk config files, so you can compare > > additionally I can send both red5sip and OM, but I need some place like > dropbox for this > > > > > > On 23 July 2014 20:51, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > OK I will down load this evening and see what happens.. > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Tuesday, July 22, 2014 11:28 PM > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > Hello Horace, > > > > just have checked, 3.0.3 seems to work as expected (at least 'SIP > Transport' sitting in the room) > > There are some NPEs in logs (will take a looks at it as soon as will have > some time) > > > > On 23 July 2014 12:13, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Ok thanks Maxim > > > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Tuesday, July 22, 2014 7:17 AM > > > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > I'll try to find server with configured Asterisk and try to double-check > > > > On 22 July 2014 20:37, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Thanks I made the change prior to sending the email. There appears to be > something else missing: > > There appears to be a entry missing in the /etc/asterisk/func_odbc.conf: > file for ${EXTEN} > > I am probably wrong. But I can’t figure out how this is making the call > to the database. > > I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file > and I am not sure how to construct one there that would work. > > Would I simply add > > [EXTEN] > dsn=asterisk > readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO > GET THE ROOMID INTO THIS VARIABLE > > > > Thanks ahead of time > > > > Miles > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Tuesday, July 22, 2014 6:23 AM > > > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > yes, this line need to be corrected > > openmeetings/rooms -> openmeetings/room > > > > guess this is the problem > > > > On 22 July 2014 19:43, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Thanks Maxim, > > I have been trying to figure this out, I am knew to it all but on a steep > learning curve. > > > > I do have a question about the asterisk extensions.conf > > exten => > _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail) > > > > Does the above line check the openmeetings database rooms table for the > confno and returns ok if it finds it and notavail if it doesn’t? > > > > I am getting the following warning: > > Chan_sip.c.:25184 handle_request_infite: Call from ‘red5sip_user’ ( > 98.0.0.0:5070) to extension ‘40016’ rejected because extension not found > in context “rooms-red5sip” I don’t recall seeing this error before. But > if the “exten” line is checking the database openmeetings and looking for > rooms table it does not exist. There is a table name room but no rooms. > > > > Am I reading this correctly? > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Tuesday, July 22, 2014 4:27 AM > > > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > AFAIK dial in/out conference room was working as expected > > Not sure if we still have infrastructure test current version > > > > Will try to ask someone > > > > On 21 July 2014 20:17, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Ok on the sip transport, I will try to figure out why it keep popping in > and out. > > However, I am not understanding concerning the Asterisk config. The > asterisk config I am using is from the install and modification as stated > from the VOIP/SIP 3.0 integration package. The instructions don’t say a > sip trunk or outside provider is required. However, I am unable to > succesfull make calls from a conference room to a phone and visa versa. > Which I thought I was suppose to be able to do after the integration. Per > the below instructions I guess I am asking what am I missing from below? > > *Feature Matrix:* > > *Feature* > > *Description* > > 1) Dial-In > > A phone number is provided which you can give to anybody to "Dial-In" via > usual landlane/phone into the conference room of OpenMeetings - Every room > has its own phone number. Currently room gets number > like 400<Id of room>. Maybe should move phone prefix to settings, > currently it hardcoded. > > 2) Dial-Out > > The users in the conference room can call anybody outside of the > conference room by entering the phone number in the conference room - In > room actions menu exist "SIP dialer". When user clicked dialer window > appears. > Currently calls can't be dropped from Openmeetings, tbd > > 3) Multiple Dial-In > > You can give away multiple numbers and do the same as described in case > (1). Multiple Dial-In is achieved by configuring the SIP-server (Asterisk). > It is possible to create multiple extensions (phone numbers) in Asterisk > configuration that will be redirects to single conference room. > > 4) Multiple Dial-Out > > You can dial multiple numbers from within the conference room - From > within conference can be dialed multiple numbers. > > *Main difference to native Red5-Phone project* > > > > > > > > *rom:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Monday, July 21, 2014 5:36 AM > > > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > If I do remember correctly > > SIP transport should enter the room and be in room as long as there other > users in it. > > > > Possibility to call to phone numbers depends on your Asterisk config. > > > > I'll try to fix documentation ASAP > > > > On 21 July 2014 18:55, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Thanks Maxim, > > Let me make sure I understand about the sip transport. It should not be > popping in and out of the room? On my box I keep getting Sip Transport has > exited the room. > > > > When properly configured, should I be able to call land and cell phones > without a need for another server or; > > 1. Do I need to subscribe to a VOIP service provider > > 2. Configure Asterisk as a sip trunk to use google voice or some > other solution? > > Also if you could have someone correct this line in the instructions of > extensions.conf it will help to eliminate at least one error > > > > > *[rooms-red5sip]exten => > _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavil) > <<<<<<<<<<<< should be “notavail”exten => > _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)exten => > _400X!,n(notavail),Hangup * > > > > *Thanks ahead of time* > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Monday, July 21, 2014 4:00 AM > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > Hello Horace, > > > > jsvc can be used to start java application as service > > I don't really like it (it was unstable when I used it) > > I prefer to write init.d script > > > > I see no errors in your log > > If everything is OK SIP transport should be in the room > > All 3 logs should be checked to have no errors > > I usually run asterisk in debug mode while setting everything up > > > > > > > > On 20 July 2014 23:55, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Additonally the VOIP and SIP integration 3.0 instructions do not mention > installing jsvc. Is it still a requirement to install jsvc under 3.0 as it > was under 2.0?: > > *apt-get install jsvc* > > > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Friday, July 18, 2014 11:18 AM > > > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > will try to take a look a look at it tomorrow, too late here ... > > > > On 19 July 2014 00:59, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Ok I will restart red5sip service and red5 and then send a new log > > > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Friday, July 18, 2014 10:06 AM > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > got the full trace in other email, will try to check code > > > > On 18 July 2014 23:22, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Did miss understand what you were asking for? > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Friday, July 18, 2014 8:56 AM > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > would be more helpful to get full stack instead of "Red5sip log : Error > o.z.s.p.SipProvider: java.lang.NullPointerException: Null" > > > > On 18 July 2014 22:37, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Openmeetings log says confBridgeList authentication is failing. I will > check to make sure I didn’t change a password there.. > > Red5sip log : Error o.z.s.p.SipProvider: java.lang.NullPointerException: > Null > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Friday, July 18, 2014 8:43 AM > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > this "I have a sip transport that keeps popping in and out of the room." > usually mean something configured wrong. > > Any exceptions in the logs (openmeetings.log and red5sip.log > > > > On 18 July 2014 22:15, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Sorry also Asterisk 11 > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Friday, July 18, 2014 8:10 AM > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > Additionally, what version are you using? > > > > On 18 July 2014 21:52, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Probably not, since I just went into a public room.. let me create a room.. > > > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Friday, July 18, 2014 8:07 AM > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > Do you have "*Enable SIP transport in the room*" checked for the room you > are testing? > > > > On 18 July 2014 21:48, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > Maxim thanks for the reply, I went back and rechecked my setup. I have > completed all the steps according to the integration document. > > I found the following document on the wiki: > https://cwiki.apache.org/confluence/display/OPENMEETINGS/VoIP+Integration+General+Description > > > > According to this document I should get a sip dialer under the rooms > actions menu. But I have no dialer there. > > > > The only error I see in the red5sip window is > > 18 Jul 07:50:11 . [nioProcessor-2]:[INFO ] > o.r.c.n.r.BaseRTMPClienthandler: No Service provider / method not found; to > handle calls like onBWCheck, add a service provider. (it is my > understanding this error is to be expect as it is not being used?) > > > > So where would I start to try and figure out why there is no sip dialer > available? > > > > > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Friday, July 18, 2014 7:15 AM > *To:* Openmeetings user-list > *Subject:* Re: VOIP and Sip Integration > > > > http://openmeetings.apache.org/voip-sip-integration.html > > > > On 18 July 2014 20:52, Horace Miles <horace.mi...@myit-solutions.com> > wrote: > > It is nice that Openmeetings provided a way to integrate VOIP and Sip with > Asterisk. That being said, I can find no documentation that tells the > following: > > If the integration was successful? > > What icons should show up where etc. > > What actions can be taken by an admin or a user for that matter i.e. how > to make a phone call out or in. > > Did I miss something somewhere? > > > > Miles > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > > > > > > -- > WBR > Maxim aka solomax > -- WBR Maxim aka solomax
