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
