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: [email protected] | p: (+34) 986 911 644 2016-07-07 10:20 GMT+02:00 Maxim Solodovnik <[email protected]>: > please use master for 3.1.x > > On Thu, Jul 7, 2016 at 1:56 PM, Elena Darriba <[email protected]> > 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: >> [email protected] | p: (+34) 986 911 644 >> >> 2016-07-06 18:13 GMT+02:00 Maxim Solodovnik <[email protected]>: >> >>> 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 <[email protected] >>> > 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: >>>> [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 >>>>> >>>> >>>> >>> >>> >>> -- >>> WBR >>> Maxim aka solomax >>> >> >> > > > -- > WBR > Maxim aka solomax >
