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
