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> 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 | p: (+34) 986 911 644 > > 2016-07-06 16:12 GMT+02:00 Maxim Solodovnik <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> >> 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 | p: (+34) 986 911 644 >>> >>> 2016-05-14 9:58 GMT+02:00 Maxim Solodovnik <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> 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 | p: (+34) 902 999 465 >>>>> >>>>> 2016-04-25 15:23 GMT+02:00 Maxim Solodovnik <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> 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 | p: (+34) 902 999 465 >>>>>>> >>>>>>> 2016-04-25 9:52 GMT+02:00 Maxim Solodovnik <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> 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 | p: (+34) 902 999 465 >>>>>>>>> >>>>>>>>> 2016-04-22 23:13 GMT+02:00 Christos Moustakakis < >>>>>>>>> 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 | 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
