Re: Status of SIP Integration in OM 6.00

I believe I've answered in other thread (I would recommend keeping the mail-thread number reasonable : This way it will be easier to follow :) On Tue, 24 Aug 2021 at 08:39, Ali Alhaidary wrote: > Hi, > > I am sorry, but it has been some time and relay do not recall the details > as we aband

Re: SIP Integration

On Sun, 22 Aug 2021 at 01:06, Yah's Global Kingdom wrote: > OK, I am able to register devices and call anything within the internal > context. But I can not dial a conference room. Can anyone that is able to > dial a conference from an Asterisk instance please share their Sip.conf and > Extensi

Re: Status of SIP Integration in OM 6.00

Hi, I am sorry, but it has been some time and relay do not recall the details as we abandoned using SIP in our production system long time ago. Ali On 8/24/21 3:51 AM, Yah's Global Kingdom wrote: Hi Ali Alhaidary, Is it possible that you can tell me were you got the settings for openmeetin

Re: Status of SIP Integration in OM 6.00

Hi Ali Alhaidary, Is it possible that you can tell me were you got the settings for openmeetings.settings in asterisk? Guess what I am asking the below settings are required to be set in openmeetings.settings file, but I don't know what they actually equate to in Asterisk. Asterisk: Where are thes

Re: SIP Integration

OK, I am able to register devices and call anything within the internal context. But I can not dial a conference room. Can anyone that is able to dial a conference from an Asterisk instance please share their Sip.conf and Extension.conf so I can compare...? On Wed, Aug 18, 2021 at 12:10 AM Maxim

Re: SIP Integration

`sudo netstat -taupen|grep aster` lists port 5060 for me On Wed, 18 Aug 2021 at 06:38, Yah's Global Kingdom wrote: > The SIP protocol uses port 5060, according to the documentation: SIP > Config tcpenble =yes and tcpbindaddress default port number is 5060. > > On Mon, Aug 16, 2021 at 9:3

Re: SIP Integration

The SIP protocol uses port 5060, according to the documentation: SIP Config tcpenble =yes and tcpbindaddress default port number is 5060. On Mon, Aug 16, 2021 at 9:34 PM Maxim Solodovnik wrote: > > > On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom > wrote: > >> Please disregard, I have gott

Re: SIP Integration

On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom wrote: > Please disregard, I have gotten the sip transport to enter the room. > However, I don't see anything in Asterisk for when the Transport agent > enters the room or when I try to register a client. > You should "see something in Asterisk"

Re: SIP Integration

Please disregard, I have gotten the sip transport to enter the room. However, I don't see anything in Asterisk for when the Transport agent enters the room or when I try to register a client. I have nothing listening on ports 5060,5061 or 5062. On Sat, Aug 14, 2021 at 11:29 AM Yah's Global Kingdo

Re: SIP Integration

Hello, I would use this https://openmeetings.apache.org/AsteriskIntegration.html guide (I followed it while I wrote it) The guide at Confluence should be outdated, please ignore it On Sun, 15 Aug 2021 at 01:29, Yah's Global Kingdom wrote: > Update: > Asterisk is not listening on ports 5060/506

Re: SIP Integration

Update: Asterisk is not listening on ports 5060/5061/5062 although I have updated the sip.conf I am using the guide at https://openmeetings.apache.org/AsteriskIntegration.html to implement Asterisk and VOIP. Before under previous additions, when I entered the room, the SIP transport agent would a

SIP Integration

I am using the guide at https://openmeetings.apache.org/AsteriskIntegration.html to implement Asterisk and VOIP. Before under previous additions, when I entered the room, the SIP transport agent would also enter the room. Now after upgrading from 5.0 to 6.10 when I enter the room no sip transport

Re: SIP Integration in OM | video support question

Well, I had some difficulties to add video support: absence or presence of video should be determined in SIP stream I guess this is doable, unfortunately I had too few free time right now :((( On Thu, 3 Jun 2021 at 13:29, Viktor Kotliar wrote: > Hi all! > > Sorry if I missed such info but are

Re: SIP Integration in OM 6.00 problem

cool! I wrote something that works :))) On Thu, 3 Jun 2021 at 12:14, Viktor Kotliar wrote: > Hi, all! > > Looks like in our case the problem was with multiple network interfaces > (after docker install). > After installing standalone kurento media server and disabling all > interfaces except eth

SIP Integration in OM | video support question

Hi all! Sorry if I missed such info but are there any plans to add video support for SIP with OM through Kurento media server? Best regards Victor

Re: SIP Integration in OM 6.00 problem

Hi, all! Looks like in our case the problem was with multiple network interfaces (after docker install). After installing standalone kurento media server and disabling all interfaces except eth0 SIP started to work two way Here was a debug message[1] which helped to find the source of the pro

SIP Integration in OM 6.00 problem

Hi all! We followed instruction from https://openmeetings.apache.org/AsteriskIntegration.html about installation of SIP and it almost works. We have SIP client1<->SIP client2 in one room can speak and hear SIP client->WEB client in one room one way only ( we do not hear at SIP anything fro

Re: SIP Integration in OM 6.00 | setup instructions

Most recent docs are here: https://openmeetings.apache.org/AsteriskIntegration.html :) On Thu, 22 Apr 2021 at 16:10, Viktor Kotliar wrote: > > Hi all, > is there anywhere fresh instructions about installation\integration SIP > with OM 6.x > > Many thx in advance! > > Best regards > Victor --

SIP Integration in OM 6.00 | setup instructions

Hi all, is there anywhere fresh instructions about installation\integration SIP with OM 6.x Many thx in advance! Best regards Victor

Re: Status of SIP Integration in OM 6.00

I would think that the conferencing bridge would be able to handle many to one connections as that is the ultimate purpose conferencing...I will have to check it out next week, right now I am in a temp location and have to move back into my office and set everything back up again. I am hoping that

Re: Status of SIP Integration in OM 6.00

Yes, running the latest build of 6.1.0, however it was long and complex learning experience for us, and, it depends on the subscription and number of lines offered, and, for some reason that I did not dig too much into, server is overloaded, and unfortunately, in our part of the globe, no opera

Re: Status of SIP Integration in OM 6.00

Thanks Ali, I have to look at my notes it has been a minute. I see emails that you are running 6.+ Are multiple user able to call in to your room conference using SIP? On Thu, Apr 15, 2021 at 2:04 PM Ali Alhaidary wrote: > upgrading is a long process, but certainly worth each and every minute

Re: Status of SIP Integration in OM 6.00

upgrading is a long process, but certainly worth each and every minute :-) Ali On 4/15/21 11:40 PM, Yah's Global Kingdom wrote: Ok thanks I will upgrade and test On Tue, Apr 13, 2021 at 5:29 PM Maxim Solodovnik > wrote: I believe so :) But my testing abil

Re: Status of SIP Integration in OM 6.00

Ok thanks I will upgrade and test On Tue, Apr 13, 2021 at 5:29 PM Maxim Solodovnik wrote: > I believe so :) > But my testing abilities are limited :) > > And SIP-video is not implemented > > from mobile (sorry for typos ;) > > > On Wed, Apr 14, 2021, 05:11 Yah's Global Kingdom > wrote: > >> Max

Re: Status of SIP Integration in OM 6.00

I believe so :) But my testing abilities are limited :) And SIP-video is not implemented from mobile (sorry for typos ;) On Wed, Apr 14, 2021, 05:11 Yah's Global Kingdom wrote: > Maxim what exactly does that mean, Can more than one person, call into > the room SIMULTANEOUSILY using SIP? > >

Re: Status of SIP Integration in OM 6.00

Maxim what exactly does that mean, Can more than one person, call into the room SIMULTANEOUSILY using SIP? On Sat, Apr 10, 2021 at 7:27 PM Maxim Solodovnik wrote: > Hello, > > According to my tests 2-way audio-only SIP should work as expected :) > > from mobile (sorry for typos ;) > > > On Thu,

Re: Status of SIP Integration in OM 6.00

Hello, According to my tests 2-way audio-only SIP should work as expected :) from mobile (sorry for typos ;) On Thu, Apr 8, 2021, 04:20 Yah's Global Kingdom wrote: > Has this been implemented yet? >

Status of SIP Integration in OM 6.00

Has this been implemented yet?

Re: SIP integration

Please see inline (from mobile, sorry for typos) On Sat, May 2, 2020, 15:07 Rohrbach, Gerald wrote: > Sebastian, > > > > is for M5 the SIP integration planned? > It depends We have lots of issues reported So maybe it worth to fix them and release 5.0.0 (without M) Then foc

SIP integration

Sebastian, is for M5 the SIP integration planned? We are using M4 already. In our case only for internal use, so it´s not that critical. And up to know it seems to be stable. The dial in is a feature often requested from users, that do have a web cam or microphone. In theory kurunto should

Re: Open Meetings 4.0.10 - Sip integration

PC speakers but the mic plays in the sip conference. Sent from Outlook Mobile<https://aka.ms/blhgte> From: Maxim Solodovnik Sent: Monday, April 6, 2020 8:59:53 PM To: Openmeetings user-list Subject: Re: Open Meetings 4.0.10 - Sip integration Accord

Re: Open Meetings 4.0.10 - Sip integration

et.rtmp.RTMPMinaIoHandler.messageReceived #138 > > 06 Apr 22:24:50 - [NioProcessor-12]:[DEBUG] o.r.c.n.r.RTMPMinaIoHandler: > messageSent > > 06 Apr 22:24:50 - [NioProcessor-12]:[DEBUG] o.r.c.n.r.RTMPConnManager: > Getting connection by session id: JARJEMMY5JTZG > > 06 Apr 22:24:50 -

RE: Open Meetings 4.0.10 - Sip integration

TZG 06 Apr 22:24:50 - [NioProcessor-12]:[DEBUG] o.r.s.n.r.RTMPConnection: Connection calls pending: 1 From: Chris Botts [mailto:cbo...@syndeonetwork.com] Sent: Monday, April 6, 2020 1:37 PM To: user@openmeetings.apache.org Subject: Open Meetings 4.0.10 - Sip integration Greetings, I have inst

Open Meetings 4.0.10 - Sip integration

Greetings, I have installed Openmeetings 4.0.10 Setup sip integration, setup https and a certificate. I am able to access the web log and all works there. I am able to call and join a conference with the room and pin. The problem I am facing is the open meeting web and the asterisk conference

Re: Sip-Integration Question.

Hi Fabian, It's generally possible, but is not realized now. Could you please file a bug in JIRA with your suggestions? Or if you need this quickly you can e-mail to our commercial support service: om.uni...@gmail.com Thank you, Irina Это сообщение было отправлено с неинфицированного компьютер

Sip-Integration Question.

Hello, I have successfully integrated Asterisk with Openmeetings, so a User has to call an external Number and inputs the conferenceroomnumber and is inside the conference. But there is one thing I would like to add to make it easier for our users, to automatically enable SIP-Transport in a

Re: SIP Integration

Hello Horace, sorry for keeping silence was busy on my daytime job :( unfortunately this is not implemented, you can fork the project on github and try to fix it. On Mon, Nov 16, 2015 at 6:56 PM, Horace Miles < horace.mi...@myit-solutions.com> wrote: > Hello Openmeetings Users, > > > > I still

Re: Openmeetings 3.0 SIP integration

o:* Openmeetings user-list > *Subject:* Re: Openmeetings 3.0 SIP integration > > > > Sorry for the late response, > > > > can you enable client debug (change DEPLOYMENT to be DEVELOPMENT in > web.xml) and check if there any errors while you are trying to dialing out? &

RE: Openmeetings 3.0 SIP integration

3.0 SIP integration Sorry for the late response, can you enable client debug (change DEPLOYMENT to be DEVELOPMENT in web.xml) and check if there any errors while you are trying to dialing out? Callers should be able to talk and hear Not sure what do you mean by "sample code" :( O

SIP Integration

Hello Openmeetings Users, I still consider myself to be a noob at this. Trying to rap my head around all the different technology. I am running Openmeetings 3.0.2. release on Ubuntu 12.04 LTS Core 2 Quad CPU Q6700 @ 2.66GHz x4 32 bit OS I have successfully integrated SIP using the SIP

Re: Openmeetings 3.0 SIP integration

Sorry for the late response, can you enable client debug (change DEPLOYMENT to be DEVELOPMENT in web.xml) and check if there any errors while you are trying to dialing out? Callers should be able to talk and hear Not sure what do you mean by "sample code" :( On Sun, Nov 1, 2015 at 9:29 PM, Horace

Re: SIP integration question - Openmeetings 3.0

Hello what version of OM are you using? can you ensure using Admin you have SIP number assigned to the room? On Mon, Nov 9, 2015 at 11:27 PM, Lucas Kay wrote: > Hello, > > Trying to configure Asterisk SIP integration following the guide at: > http://openmeetings.apache

SIP integration question - Openmeetings 3.0

Hello, Trying to configure Asterisk SIP integration following the guide at: http://openmeetings.apache.org/red5sip-integration_3.0.html but I am so far unsuccessful. Below is the log. Any help would be greatly appreciated. 02 lis 16:34:18 - [main]:[INFO ] o.r.c.n.r.BaseRTMPClientHandler: rtmp

Openmeetings 3.0 SIP integration

I can get into the conference room just fine. However dialing out does not work. Also should callers that have dialed in on their phone be able to talk in the conference or listen only? Does anyone have any sample code to dial out of conference rooms in Openmeeting? --- This email has bee

Re: Sip integration

please do not write direct emails I'm not sure I can understand your question red5sip performs audio/video transcoding between OM and Asterisk I never tried to have all 3 on different servers On Wed, Sep 23, 2015 at 8:28 PM, Awotipe Oluwaseun < awotipeoluwas...@yahoo.com> wrote: > Hello, > Thank

Re: Sip integration

Hello Awotipe, you need to use same DB for OM and Asterisk hightlighted part is sip-username, the user in Asterisk having rights to modify the database On Mon, Sep 14, 2015 at 9:33 PM, awotipe oluwaseun < awotipeoluwas...@yahoo.com> wrote: > hi maxim, > i do have some specific question, at the a

Re: Sip integration

any updates to the SIP integration since the 3.0 > instructions were put out. i.e. any way to kick users that are connected > via sip etc? > > > > *From:* Maxim Solodovnik [mailto:solomax...@gmail.com] > *Sent:* Saturday, September 05, 2015 9:40 PM > *To:* Awotipe Oluwaseun &g

Re: Sip integration

hi maxim,i do have some specific question, at the asterisk connector config below[asterisk-connector] Description = MySQL connection to 'openmeetings' database Driver = MySQL Database = openmeetings Server = localhost USER = root PASSWORD = Port = 3306 Socket = /var/run/mysqld/mysqld.sock where da

RE: Sip integration

Maxim, Have there been any updates to the SIP integration since the 3.0 instructions were put out. i.e. any way to kick users that are connected via sip etc? From: Maxim Solodovnik [mailto:solomax...@gmail.com] Sent: Saturday, September 05, 2015 9:40 PM To: Awotipe Oluwaseun Cc: Openmeetings

RE: Sip integration

Asterisk does not use the openmeetings database, it uses it’s own database that is installed when you install asterisk From: Pierre Smits [mailto:pierre.sm...@gmail.com] Sent: Thursday, September 03, 2015 1:10 PM To: user@openmeetings.apache.org Subject: Fwd: Sip integration FYI To keep

Re: Sip integration

To be fair, I don't remember :((( I performed configuration long ago, but I did exactly what is written on the configuration page, step by step :)) I can check existing installation if you have any particulal question :) On Fri, Sep 4, 2015 at 8:24 PM, Awotipe Oluwaseun < awotipeoluwas...@yahoo.co

Re: Sip integration

These steps: Update Openmeetings with creadentials for Asterisk manager. Modify /opt/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-applicationContext.xml find ** uncomment its parameters and set it to your custom values. IMPORTANT: this step should be done *BEFORE* system install/restor

Re: Sip integration

Maybe Asterisk connection on OM side wasn't properly configured? On Fri, Sep 4, 2015 at 4:01 AM, awotipe oluwaseun < awotipeoluwas...@yahoo.com> wrote: > thanks smith for your last comment and guildiance, i have some more othaer > questions from the instruction at > >

Re: Sip integration

thanks smith for your last comment and guildiance, i have some more othaer questions from the instruction at                                      Modify  /etc/asterisk/extensions.conf                                ; *; The below dial plan is u

Fwd: Sip integration

eun Date: Thu, Sep 3, 2015 at 10:08 PM Subject: Re: Sip integration To: Pierre Smits Thanks for your reply, one more questions the database that was being refer to in the prelim-screen shot, is asterisk my SQL or openmeetings database setup initially. oluwaseun On Sep 3, 2015 10:54 AM, Pierr

Re: Sip integration

I wouldn't recommend it for production purposes. There it is best to have separate setups for OpenMeetings, the RDBMS and the SIP server. But for evaluation purposes an all-in-one box could suffice. As the document states in the section below OEM-prelim-ScreenShot-product the following line determ

Re: VOIP and Sip Integration

pears 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 u

RE: VOIP and Sip Integration

...@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

Re: VOIP and Sip Integration

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

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 abl

Re: VOIP and Sip Integration

; *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

Re: VOIP and Sip Integration

t; > > > *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 > > &

Re: VOIP and Sip Integration

x...@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 > >

Re: VOIP and Sip Integration

r 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] > *S

Re: VOIP and Sip Integration

; > *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 re

Re: VOIP and Sip Integration

lt_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 Integra

RE: VOIP and Sip Integration

me 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 pr

Re: VOIP and Sip Integration

ually run asterisk in debug mode while setting everything up On 20 July 2014 23:55, Horace Miles 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?: > &g

RE: VOIP and Sip Integration

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

RE: VOIP and Sip Integration

where do I check? 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 wrote: Ok I

Re: VOIP and Sip Integration

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 > wrote: > > Did miss understand what you were asking for

RE: VOIP and Sip Integration

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

Re: VOIP and Sip Integration

> 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 >> >>

Re: VOIP and Sip Integration

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 > wrote: > &

RE: VOIP and Sip Integration

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 :

Re: VOIP and Sip Integration

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

RE: VOIP and Sip Integration

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 Mi

Re: VOIP and Sip Integration

olodovnik [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 > wrote: > >

RE: VOIP and Sip Integration

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 wrote: Probably not, since I

RE: VOIP and Sip Integration

: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 wrote: Probably not, since I just went into a public room.. let me create a room.. From: Maxim Solodovnik [mailto:solomax...@gmail.com

Re: VOIP and Sip Integration

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 > wrote: > > Maxim thanks fo

RE: VOIP and Sip Integration

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&quo

Re: VOIP and Sip Integration

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

RE: VOIP and Sip Integration

: 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 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

Re: VOIP and Sip Integration

http://openmeetings.apache.org/voip-sip-integration.html On 18 July 2014 20:52, Horace Miles 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 succe

VOIP and Sip Integration

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

Re: starting red5sip for SIP integration

The "jsvc error: JSVC re-exec requires execution with an absolute or relative path" error was fixed by changing jsvc to /usr/bin/jsvc in red5sip..sh This was on CentOS 6.3 BC On 01/31/13 13:00, Maxim Solodovnik wrote: I'll ask your questions the guy who wrote/fix red5sip and will let you kn

Re: starting red5sip for SIP integration

I'll ask your questions the guy who wrote/fix red5sip and will let you know On Thu, Jan 31, 2013 at 6:52 PM, Bart Coninckx wrote: > All, > > Am trying to get the Asterisk integration done and I stumble into a > problem while starting red5sip with the red5sip.sh script. > > I changed the JAVA_HOM

starting red5sip for SIP integration

All, Am trying to get the Asterisk integration done and I stumble into a problem while starting red5sip with the red5sip.sh script. I changed the JAVA_HOME variable in it as well as the LOGS_DIR which pointed wrongly to "logs" while the svn checkout produces "log". However, when running the