Hi, i'm compiling right now version 3.1.1 and as stated in http://openmeetings.apache.org/BuildInstructions.html you have to use maven instead of ant.
BTW I don't know if its fixed in trunk, but i have edited pom.xml
because apache rat plugin snapshot is not available and changed line
917 to use version 0.,12:
<groupId>org.apache.rat</groupId>
<artifactId>apache-rat-plugin</artifactId>
<version>0.12</version>
<configuration>
Kind regards
El 12/07/16 a las 13:56, Elena Darriba escribió:
> Hello Maxim,
>
> Sorry, I am trying compiling master but there is not build.xml file...
> how could I proceed?
>
> [root@centos7-1 red5sip]# git checkout master
> Already on 'master'
> [root@centos7-1 red5sip]# ls -l
> total 8
> drwxr-xr-x 2 root root 24 jul 12 13:52 lib
> drwxr-xr-x 2 root root 24 jun 29 10:22 log
> drwxr-xr-x 4 root root 30 jun 29 09:57 out
> -rw-r--r-- 1 root root 3577 jul 12 13:52 pom.xml
> -rw-r--r-- 1 root root 123 jul 12 13:52 README.md
> drwxr-xr-x 3 root root 17 jul 12 13:52 src
> [root@centos7-1 red5sip]# ant
> Buildfile: build.xml does not exist!
> Build failed
> [root@centos7-1 red5sip]#
>
>
> Thanks in advance.
>
> Best Regards,
> Elena.
>
> *Elena Darriba*
> VoIP Systems Engineer @ Quobis <http://www.quobis.com/> |
> e: elena.darr...@quobis.com <mailto:elena.darr...@quobis.com> |
> p: (+34) 986 911 644
>
> 2016-07-07 10:20 GMT+02:00 Maxim Solodovnik <solomax...@gmail.com
> <mailto:solomax...@gmail.com>>:
>
> please use master for 3.1.x
>
> On Thu, Jul 7, 2016 at 1:56 PM, Elena Darriba
> <elena.darr...@quobis.com <mailto:elena.darr...@quobis.com>> wrote:
>
> Hi Maxim,
>
> The current version is 3.1.x, sorry for my mistake.
>
> What branch of red5sip must I to compile?
>
> Thanks in advance.
>
> BR,
> Elena.
>
> *Elena Darriba*
> VoIP Systems Engineer @ Quobis <http://www.quobis.com/> |
> e: elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com> | p: (+34) 986 911 644
>
> 2016-07-06 18:13 GMT+02:00 Maxim Solodovnik
> <solomax...@gmail.com <mailto:solomax...@gmail.com>>:
>
> I recently have fixed master branch to be buildable (for
> 3.1.x)
> Do you have any particular reason to use 3.0.x instead of
> 3.1.x?
>
>
> On Wed, Jul 6, 2016 at 10:00 PM, Elena Darriba
> <elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com>> wrote:
>
> Hello Maxim,
>
> We are using OM 3.0 on CentOS 7.2, I will check the
> exact version of OM. What branch should I compile?
>
> Thanks.
>
> BR,
> Elena.
>
> *Elena Darriba*
> VoIP Systems Engineer @ Quobis
> <http://www.quobis.com/> | e: elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com> | p: (+34) 986 911 644
>
> 2016-07-06 16:12 GMT+02:00 Maxim Solodovnik
> <solomax...@gmail.com <mailto:solomax...@gmail.com>>:
>
> Hello Elena,
>
> https://github.com/openmeetings/red5sip/tree/red5sip_3.0
> branch is stored for historical reasons and might
> be not compilable
> what version of OM are you using?
>
>
> On Wed, Jun 29, 2016 at 7:16 PM, Elena Darriba
> <elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com>> wrote:
>
> Hello Maxim,
>
> I tried to compile src code from master (using
> red5sip_3.0 branch) and I detected some
> errors. How can I compile the new version on
> master?
>
> Thanks in advance.
>
> Best Regards,
> Elena.
>
> *Elena Darriba*
> VoIP Systems Engineer @ Quobis
> <http://www.quobis.com/> |
> e: elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com> | p: (+34)
> 986 911 644
>
> 2016-05-14 9:58 GMT+02:00 Maxim Solodovnik
> <solomax...@gmail.com
> <mailto:solomax...@gmail.com>>:
>
> Hello Elena,
>
> I have finally installed Asterisk
> fixed
> red5sip:
> https://github.com/openmeetings/red5sip/tree/master
> hopefully will be able to test everything
> together (hopefully LinPhone will work
> with Asterisk)
>
> On Thu, May 12, 2016 at 12:48 PM, Elena
> Darriba <elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com>> wrote:
>
> Hello Maxim,
>
> Have you any update about this issue?
>
> Thanks in advance.
>
> Best Regards,
> Elena.
>
> *Elena Darriba*
> VoIP Systems Engineer @ Quobis
> <http://www.quobis.com/> |
> e: elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com> |
> p: (+34) 902 999 465
>
> 2016-04-25 15:23 GMT+02:00 Maxim
> Solodovnik <solomax...@gmail.com
> <mailto:solomax...@gmail.com>>:
>
> will try to do it this week
>
> @Timur, maybe you can help?
>
> On Mon, Apr 25, 2016 at 5:59 PM,
> Elena Darriba
> <elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com>>
> wrote:
>
> Hello Maxim,
>
> Please, could you tell me an
> aproximate date for this review?
>
> Thanks in advance,
>
> Best Regards,
> Elena.
>
> *Elena Darriba*
> VoIP Systems Engineer @ Quobis
> <http://www.quobis.com/> |
> e: elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com> |
> p: (+34) 902 999 465
>
> 2016-04-25 9:52 GMT+02:00
> Maxim Solodovnik
> <solomax...@gmail.com
> <mailto:solomax...@gmail.com>>:
>
> Hello All,
>
> sorry for keeping silence
> on the topic,
> Unfortunately I had no
> time to configure asterisk
> server (old one deceased)
> I'll write back as soon as
> I'll find time and check
> the issue
>
> On Mon, Apr 25, 2016 at
> 1:29 PM, Elena Darriba
> <elena.darr...@quobis.com
> <mailto:elena.darr...@quobis.com>>
> wrote:
>
> Dear Christos,
>
> Install Asterisk is
> very easy, you can
> compile the code so
> you can use Debian,
> Ubuntu or other OS.
> Also I think you can
> download it from
> repositories. I use
> the following
> instructions:
> http://openmeetings.apache.org/red5sip-integration_3.0.html
>
> Then, when Asterisk
> and red5sip are
> running, you can set
> users and create a SIP
> room in OpenMeetings.
> In my scenario, SIP
> signaling is OK, and
> users can use SIP
> room, but there is
> uncomfortable noise
> and in some cases is
> impossible to listen
> the other caller party.
>
> Thanks,
>
> Best Regards,
> Elena.
>
> *Elena Darriba*
> VoIP Systems Engineer
> @ Quobis
> <http://www.quobis.com/> |
> e: elena.darr...@quobis.com
>
> <mailto:elena.darr...@quobis.com> |
> p: (+34) 902 999 465
>
> 2016-04-22 23:13
> GMT+02:00 Christos
> Moustakakis
> <moustaka...@yahoo.gr
>
> <mailto:moustaka...@yahoo.gr>>:
>
> Dear Elena,
>
> Could you please
> send me the
> instructions you
> follow to install
> the Asterisk in
> Debian because I
> have tried to
> install in Ubuntu
> 14.04 and I didn't
> manage?
>
> Also, I would like
> to ask, when
> someone install
> the Asterisk could
> set any sip account?
>
> Thanks.
> Christos.
>
>
>
>
> Hello:
>
> We have an
> scenario with
> OpenMeetings 3,
> red5sip and
> Asterisk installed
> on a Debian
> following the
> official
> instructions. SIP
> signaling is
> correct and calls
> established
> normally, but
> users listen noise
> during a call and
> sometimes is
> impossible to hear
> the other caller
> party.
>
> We are carrying
> tests using
> FreeSWITCH on
> different OS
> (RHEL, CentOS)
> instead Asterisk
> and also using
> older versions of
> OM but results are
> the same.
>
> RTP captured
> between Asterisk
> and Red5SIP sounds
> without noise.
>
> Does anybody faced
> a situation like
> this? Could you
> please help us?
>
> Thanks in advance.
>
> Best Regards,
> Elena.
>
> *Elena Darriba*
> VoIP Systems
> Engineer @ Quobis
> <http://www.quobis.com/> |
> e:
> elena.darr...@quobis.com
>
> <mailto:elena.darr...@quobis.com> |
> p: (+34) 902 999 465
>
>
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
>
>
>
> --
> WBR
> Maxim aka solomax
>
>
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