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
