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: [email protected] | p: (+34) 986 911 644 2016-07-06 16:12 GMT+02:00 Maxim Solodovnik <[email protected]>: > 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 <[email protected]> > 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: >> [email protected] | p: (+34) 986 911 644 >> >> 2016-05-14 9:58 GMT+02:00 Maxim Solodovnik <[email protected]>: >> >>> 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 < >>> [email protected]> 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: >>>> [email protected] | p: (+34) 902 999 465 >>>> >>>> 2016-04-25 15:23 GMT+02:00 Maxim Solodovnik <[email protected]>: >>>> >>>>> will try to do it this week >>>>> >>>>> @Timur, maybe you can help? >>>>> >>>>> On Mon, Apr 25, 2016 at 5:59 PM, Elena Darriba < >>>>> [email protected]> 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: >>>>>> [email protected] | p: (+34) 902 999 465 >>>>>> >>>>>> 2016-04-25 9:52 GMT+02:00 Maxim Solodovnik <[email protected]>: >>>>>> >>>>>>> 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 < >>>>>>> [email protected]> 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: >>>>>>>> [email protected] | p: (+34) 902 999 465 >>>>>>>> >>>>>>>> 2016-04-22 23:13 GMT+02:00 Christos Moustakakis < >>>>>>>> [email protected]>: >>>>>>>> >>>>>>>>> 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: >>>>>>>>> [email protected] | p: (+34) 902 999 465 >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> WBR >>>>>>> Maxim aka solomax >>>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> WBR >>>>> Maxim aka solomax >>>>> >>>> >>>> >>> >>> >>> -- >>> WBR >>> Maxim aka solomax >>> >> >> > > > -- > WBR > Maxim aka solomax >
