I am compiling red5sip, not red5... *Elena Darriba* VoIP Systems Engineer @ Quobis <http://www.quobis.com/> | e: elena.darr...@quobis.com | p: (+34) 986 911 644
2016-07-12 14:23 GMT+02:00 Juan Carrera <carre...@unizar.es>: > 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>elena.darr...@quobis.com | p: (+34) 986 911 644 > > 2016-07-07 10:20 GMT+02:00 Maxim Solodovnik <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>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 | p: (+34) 986 911 644 >>> >>> 2016-07-06 18:13 GMT+02:00 Maxim Solodovnik < <solomax...@gmail.com> >>> 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>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>elena.darr...@quobis.com | p: (+34) 986 911 >>>>> 644 >>>>> >>>>> 2016-07-06 16:12 GMT+02:00 Maxim Solodovnik < <solomax...@gmail.com> >>>>> solomax...@gmail.com>: >>>>> >>>>>> Hello Elena, >>>>>> >>>>>> <https://github.com/openmeetings/red5sip/tree/red5sip_3.0> >>>>>> 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>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>elena.darr...@quobis.com | p: (+34) 986 >>>>>>> 911 644 >>>>>>> >>>>>>> 2016-05-14 9:58 GMT+02:00 Maxim Solodovnik < <solomax...@gmail.com> >>>>>>> solomax...@gmail.com>: >>>>>>> >>>>>>>> Hello Elena, >>>>>>>> >>>>>>>> I have finally installed Asterisk >>>>>>>> fixed red5sip: >>>>>>>> <https://github.com/openmeetings/red5sip/tree/master> >>>>>>>> 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>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>elena.darr...@quobis.com | p: (+34) 902 >>>>>>>>> 999 465 >>>>>>>>> >>>>>>>>> 2016-04-25 15:23 GMT+02:00 Maxim Solodovnik < >>>>>>>>> <solomax...@gmail.com>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>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>elena.darr...@quobis.com | p: (+34) >>>>>>>>>>> 902 999 465 >>>>>>>>>>> >>>>>>>>>>> 2016-04-25 9:52 GMT+02:00 Maxim Solodovnik < >>>>>>>>>>> <solomax...@gmail.com>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>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> >>>>>>>>>>>>> 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>elena.darr...@quobis.com | p: (+34) >>>>>>>>>>>>> 902 999 465 >>>>>>>>>>>>> >>>>>>>>>>>>> 2016-04-22 23:13 GMT+02:00 Christos Moustakakis < >>>>>>>>>>>>> <moustaka...@yahoo.gr>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>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 >> > > > -- > [image: Logotipo del servicio de informática y comunicaciones. Universidad > Zaragoza] > > *Juan Ramón Carrera Marcén * > * Área de aplicaciones* > Residencia de profesores > Pedro Cerbuna 12, 50009 Zaragoza > Tel. (34) 876553689 > carre...@unizar.es > > [image: unizar.es] >
