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
