Do you guys have docs on how you integrated it with Freeswitch? I’d love to get that done as well.
From: Maxim Solodovnik [mailto:[email protected]] Sent: Monday, March 27, 2017 1:21 PM To: Alejandro Alonso <[email protected]> Cc: Openmeetings user-list <[email protected]>; Elena Darriba <[email protected]> Subject: Re: Red5SIP: Uncomfortable noise during a call That is no good :( this most probably mean this is red5sip issue :( Will try to find some time and reproduce this issue locally On Mon, Mar 27, 2017 at 11:48 PM, Alejandro Alonso <[email protected]<mailto:[email protected]>> wrote: Hi Maxim, I hope you are doing well. When the two users are in the same place the audio is ok. We tested calls with both of them logged in the OM room and another one with two users in softphones. The noise is only present when there is a user in the OM room and other using a softphone. Could it be caused by some audio conversion? We have assigned more resources to the host but the noise is still there. Do you have any other idea? Thanks, Alejandro Alejandro Alonso Ferreira VoIP Systems Engineer @ Quobis<http://quobis.com/> | e: [email protected]<mailto:[email protected]> | p: (+34) 902 999 465<tel:%28%2B34%29%20902%20999%20465> — This electronic message contains information which may be privileged or confidential. The information is intended to be for the use of the individual(s) or entity named above. 2016-11-07 17:08 GMT+01:00 Maxim Solodovnik <[email protected]<mailto:[email protected]>>: Do you have issues in case only OM users are in room? Can you connect to the same room as 2 free-node users (with no OM users), is there any noise in this case? On Mon, Nov 7, 2016 at 9:33 PM, Alejandro Alonso <[email protected]<mailto:[email protected]>> wrote: Hi Maxim, Thanks for your answer. The noise starts when someone else joins the conference and only the OM user hears this noise. We have used different softphones and microphones/headphones with similar results. In addition, we use this same equipment in other scenarios (Hangout, Skype) and they work fine. Best Regards, Alejandro Alejandro Alonso Ferreira VoIP Systems Engineer @ Quobis<http://quobis.com/> | e: [email protected]<mailto:[email protected]> | p: (+34) 902 999 465<tel:%28%2B34%29%20902%20999%20465> — This electronic message contains information which may be privileged or confidential. The information is intended to be for the use of the individual(s) or entity named above. 2016-11-07 14:19 GMT+01:00 Maxim Solodovnik <[email protected]<mailto:[email protected]>>: Hello Alejandro, Am I right thinking you have clear sound without softphone connected to room? Can you try to use better hands free with your softphone? According to my observations good Mic/Headphones can significantly improve situation On Fri, Nov 4, 2016 at 9:03 PM, Alejandro Alonso <[email protected]<mailto:[email protected]>> wrote: Hi Maxim, Finally we have been able to integrate OM3.1 (red5312) with FreeSWITCH without using Asterisk. The scenario has improved a bit, now if the user is registered in a softphone he can hear the conference audio correctly. On the other hand, if the user is logged in the OM he still hears a lot of noise, even when the other users are in silence. Do you have any idea about what could be causing this? Thanks in advance. Best Regards, Alejandro Alejandro Alonso Ferreira VoIP Systems Engineer @ Quobis<http://quobis.com/> | e: [email protected]<mailto:[email protected]> | p: (+34) 902 999 465<tel:%28%2B34%29%20902%20999%20465> — This electronic message contains information which may be privileged or confidential. The information is intended to be for the use of the individual(s) or entity named above. 2016-10-11 18:53 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[email protected]>>: I would say SIP part haven't changed alot the main changes are: 1) change source location 2) change build system to maven 3) fix build I can fix issue after my vacation, you can help me to reproduce it :) On Tue, Oct 11, 2016 at 10:14 PM, Elena Darriba <[email protected]<mailto:[email protected]>> wrote: Hello Maxim, Ok, thank you very much, we are doing several tests and new version does not seem to connect well with freeswitch... we are reviewing it, thanks. BR, Elena. Elena Darriba VoIP Systems Engineer @ Quobis<http://www.quobis.com/> | e: [email protected]<mailto:[email protected]> | p: (+34) 986 911 644 2016-10-11 17:08 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[email protected]>>: Hello Elena, to be fair I don't know, I believe in case it was able to integrate it before it is possible to integrate it now ... On Tue, Oct 11, 2016 at 7:45 PM, Elena Darriba <[email protected]<mailto:[email protected]>> wrote: Hello Maxim: Is it possible integrate OM3.1 (red5312) with FreeSWITCH without Asterisk? Perhaps VoIP integration is it available only with Asterisk? Older red5 versions works fine with FreeSWITCH, but the mentioned noise appeared. Thanks in advance. Best Regards, Elena Elena Darriba VoIP Systems Engineer @ Quobis<http://www.quobis.com/> | e: [email protected]<mailto:[email protected]> | p: (+34) 986 911 644 2016-07-12 15:48 GMT+02:00 Elena Darriba <[email protected]<mailto:[email protected]>>: Ok, I test it as soon as possible, thanks! Elena Darriba VoIP Systems Engineer @ Quobis<http://www.quobis.com/> | e: [email protected]<mailto:[email protected]> | p: (+34) 986 911 644 2016-07-12 15:45 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[email protected]>>: Maybe you can try latest maven? WBR, Maxim (from mobile, sorry for the typos) On Jul 12, 2016 19:30, "Elena Darriba" <[email protected]<mailto:[email protected]>> wrote: [root@centos7-1 ~]# mvn -v Apache Maven 3.0.5 (Red Hat 3.0.5-16) Maven home: /usr/share/maven Java version: 1.7.0_79, vendor: Oracle Corporation Java home: /opt/jdk1.7.0_79/jre Default locale: es_ES, platform encoding: UTF-8 OS name: "linux", version: "3.10.0-327.22.2.el7.x86_64", arch: "amd64", family: "unix" Elena Darriba VoIP Systems Engineer @ Quobis<http://www.quobis.com/> | e: [email protected]<mailto:[email protected]> | p: (+34) 986 911 644 2016-07-12 15:25 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[email protected]>>: for some weird reason default maven repository is not being tried what is your maven version? On Tue, Jul 12, 2016 at 7:22 PM, Elena Darriba <[email protected]<mailto:[email protected]>> wrote: Hi Maxim, Thanks a lot for your comments. Documentation is not updated with maven... I am trying to compile it and it shows the following error: [INFO] Scanning for projects... [INFO] [INFO] ------------------------------------------------------------------------ [INFO] Building red5sip 3.1-SNAPSHOT [INFO] ------------------------------------------------------------------------ [INFO] ------------------------------------------------------------------------ [INFO] BUILD FAILURE [INFO] ------------------------------------------------------------------------ [INFO] Total time: 6.758s [INFO] Finished at: Tue Jul 12 15:15:01 CEST 2016 [INFO] Final Memory: 9M/22M [INFO] ------------------------------------------------------------------------ [ERROR] Failed to execute goal on project red5sip: Could not resolve dependencies for project org.red5.sip:red5sip:jar:3.1-SNAPSHOT: Failed to collect dependencies for [org.red5:red5-client:jar:1.0.7-RELEASE (compile), javax.media:jmf:jar:2.1.1e (compile), commons-daemon:commons-daemon:jar:1.0.15 (compile), org.apache.openmeetings:openmeetings-db:jar:3.1.2 (compile), junit:junit:jar:4.12 (test)]: Failed to read artifact descriptor for org.apache.openmeetings:openmeetings-db:jar:3.1.2: Failure to find org.apache:apache:pom:18-SNAPSHOT in https://dl.bintray.com/openmeetings/maven was cached in the local repository, resolution will not be reattempted until the update interval of openmeetings-bintray has elapsed or updates are forced -> [Help 1] Thanks in advance. BR, Elena. Elena Darriba VoIP Systems Engineer @ Quobis<http://www.quobis.com/> | e: [email protected]<mailto:[email protected]> | p: (+34) 986 911 644 2016-07-12 15:06 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[email protected]>>: you need to build using maven: mvn clean package On Tue, Jul 12, 2016 at 6:28 PM, Elena Darriba <[email protected]<mailto:[email protected]>> wrote: I am compiling red5sip, not red5... Elena Darriba VoIP Systems Engineer @ Quobis<http://www.quobis.com/> | e: [email protected]<mailto:[email protected]> | p: (+34) 986 911 644 2016-07-12 14:23 GMT+02:00 Juan Carrera <[email protected]<mailto:[email protected]>>: Hi, i'm compiling right now version 3.1.1 and as stated in http://openmeetings.apache.org/BuildInstructions.html you have to use maven instead of ant. BTW I don't know if its fixed in trunk, but i have edited pom.xml because apache rat plugin snapshot is not available and changed line 917 to use version 0.,12: <groupId>org.apache.rat</groupId> <artifactId>apache-rat-plugin</artifactId> <version>0.12</version> <configuration> Kind regards El 12/07/16 a las 13:56, Elena Darriba escribió: 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]<mailto:[email protected]> | p: (+34) 986 911 644 2016-07-07 10:20 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[email protected]>>: please use master for 3.1.x On Thu, Jul 7, 2016 at 1:56 PM, Elena Darriba <[email protected]<mailto:[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]<mailto:[email protected]> | p: (+34) 986 911 644 2016-07-06 18:13 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[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]<mailto:[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]<mailto:[email protected]> | p: (+34) 986 911 644 2016-07-06 16:12 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[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]<mailto:[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]<mailto:[email protected]> | p: (+34) 986 911 644 2016-05-14 9:58 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[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]<mailto:[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]<mailto:[email protected]> | p: (+34) 902 999 465 2016-04-25 15:23 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[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]<mailto:[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]<mailto:[email protected]> | p: (+34) 902 999 465 2016-04-25 9:52 GMT+02:00 Maxim Solodovnik <[email protected]<mailto:[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]<mailto:[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]<mailto:[email protected]> | p: (+34) 902 999 465 2016-04-22 23:13 GMT+02:00 Christos Moustakakis <[email protected]<mailto:[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]<mailto:[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 -- [Logotipo del servicio de informática y comunicaciones. Universidad Zaragoza] Juan Ramón Carrera Marcén Área de aplicaciones Residencia de profesores Pedro Cerbuna 12, 50009 Zaragoza Tel. (34) 876553689 [email protected]<mailto:[email protected]> [unizar.es] -- WBR Maxim aka solomax -- WBR Maxim aka solomax -- WBR Maxim aka solomax -- WBR Maxim aka solomax -- WBR Maxim aka solomax -- WBR Maxim aka solomax -- WBR Maxim aka solomax In great health, Lance Oreste Telecommunications Engineer II Information Technology Department Great HealthWorks, Inc. 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