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:[email protected]] 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 <[email protected]> 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 <[email protected]> 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:[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 8:44 AM To: Horace Miles Subject: Re: VOIP and Sip Integration uploaded On 24 July 2014 20:40, Horace Miles <[email protected]> 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:[email protected]] 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 <[email protected]> wrote: OK I will down load this evening and see what happens.. From: Maxim Solodovnik [mailto:[email protected]] 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 <[email protected]> wrote: Ok thanks Maxim From: Maxim Solodovnik [mailto:[email protected]] 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 <[email protected]> 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 ${EXTEN} I am probably wrong. But I can’t figure out how this is making the call to the database. I don’t find any SQL statement in the /etc/asterisk/func_odbc.conf file and I am not sure how to construct one there that would work. Would I simply add [EXTEN] dsn=asterisk readsql=SELECT confno from room where confno = @EXTEN – NOT SURE HOW TO GET THE ROOMID INTO THIS VARIABLE Thanks ahead of time Miles From: Maxim Solodovnik [mailto:[email protected]] Sent: Tuesday, July 22, 2014 6:23 AM To: Openmeetings user-list Subject: Re: VOIP and Sip Integration yes, this line need to be corrected openmeetings/rooms -> openmeetings/room guess this is the problem On 22 July 2014 19:43, Horace Miles <[email protected]> wrote: Thanks Maxim, I have been trying to figure this out, I am knew to it all but on a steep learning curve. I do have a question about the asterisk extensions.conf exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${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:[email protected]] 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 <[email protected]> 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:[email protected]] 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 <[email protected]> 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:[email protected]] 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 <[email protected]> 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:[email protected]] 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 <[email protected]> wrote: Ok I will restart red5sip service and red5 and then send a new log From: Maxim Solodovnik [mailto:[email protected]] 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 <[email protected]> wrote: Did miss understand what you were asking for? From: Maxim Solodovnik [mailto:[email protected]] 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 <[email protected]> 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:[email protected]] 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 <[email protected]> wrote: Sorry also Asterisk 11 From: Maxim Solodovnik [mailto:[email protected]] 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 <[email protected]> wrote: Probably not, since I just went into a public room.. let me create a room.. From: Maxim Solodovnik [mailto:[email protected]] 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 <[email protected]> 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:[email protected]] 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 <[email protected]> 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
