Re: [asterisk-users] Leave and re-enter a conference
Thanks for the quick response. I will absolutely check out the ConfBridge feature. Regards; John -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Sunday, August 14, 2016 4:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Leave and re-enter a conference On Sun, Aug 14, 2016 at 1:28 PM, Tech Support wrote: > All; > > What I want to do is create a way to easily send callers into a > conference room to have an N-way conference call. I created an > extension ‘100’ that calls the MeetMe() command. Then all I need to do > is transfer a caller using a blind transfer (*2 in my case) to > extension 100. Then I can dial a feature code that sends me into that > conference (*15 in my case). So far, a piece of cake. What I realize > now is that once I enter the conference, I can’t add more people to > the call. What I need is a way to easily exit the conference, call another > user, add them to the call, etc. > and then re-enter the room myself. I tried using the ‘p’ and ‘X’ > meetme options without success. In other words: > > > > Place a call. > > Blind transfer the call to the conference (*2100) > > Enter the conference myself (*15) > > Exit the conference > > Repeat as necessary > > > > Any insight at all would be greatly appreciated. > > Thanks; > > John > This is actually where ConfBridge shines. The flexibility of ConfBridge's menu options lets you build whatever custom actions you want triggered from participants in the conference. If you use the dialplan_exec DTMF menu option [1], you can have the ConfBridge participant bounce out to the dialplan. From there, you would execute Originate to call in another participant. Note that you need to use Originate instead of Dial, as you would otherwise have the participant be bridged in a new bridge with whoever they dialed. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Leave and re-enter a conference
On Sun, Aug 14, 2016 at 1:28 PM, Tech Support wrote: > All; > > What I want to do is create a way to easily send callers into a > conference room to have an N-way conference call. I created an extension > ‘100’ that calls the MeetMe() command. Then all I need to do is transfer a > caller using a blind transfer (*2 in my case) to extension 100. Then I can > dial a feature code that sends me into that conference (*15 in my case). So > far, a piece of cake. What I realize now is that once I enter the > conference, I can’t add more people to the call. What I need is a way to > easily exit the conference, call another user, add them to the call, etc. > and then re-enter the room myself. I tried using the ‘p’ and ‘X’ meetme > options without success. In other words: > > > > Place a call. > > Blind transfer the call to the conference (*2100) > > Enter the conference myself (*15) > > Exit the conference > > Repeat as necessary > > > > Any insight at all would be greatly appreciated. > > Thanks; > > John > This is actually where ConfBridge shines. The flexibility of ConfBridge's menu options lets you build whatever custom actions you want triggered from participants in the conference. If you use the dialplan_exec DTMF menu option [1], you can have the ConfBridge participant bounce out to the dialplan. From there, you would execute Originate to call in another participant. Note that you need to use Originate instead of Dial, as you would otherwise have the participant be bridged in a new bridge with whoever they dialed. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Leave and re-enter a conference
All; What I want to do is create a way to easily send callers into a conference room to have an N-way conference call. I created an extension '100' that calls the MeetMe() command. Then all I need to do is transfer a caller using a blind transfer (*2 in my case) to extension 100. Then I can dial a feature code that sends me into that conference (*15 in my case). So far, a piece of cake. What I realize now is that once I enter the conference, I can't add more people to the call. What I need is a way to easily exit the conference, call another user, add them to the call, etc. and then re-enter the room myself. I tried using the 'p' and 'X' meetme options without success. In other words: Place a call. Blind transfer the call to the conference (*2100) Enter the conference myself (*15) Exit the conference Repeat as necessary Any insight at all would be greatly appreciated. Thanks; John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello I've succeeded in installing Asterisk 13 and more important : I can make webRTC call and I have audio !! For those on the search like myself, I want to spare some weeks of headache. My steps (CentOS 6.8) : yum install uuid-devel libuuid-devel autoconf patch automake libcurl-devel libogg-devel libvorbis-devel speex-devel popt-devel libtool-ltdl-devel libresample-devel gsm-devel libedit-devel python-devel jansson-devel binutils-devel wget http://www.pjsip.org/release/2.5.5/pjproject-2.5.5.tar.bz2 tar -xjvf pjproject-2.5.5.tar.bz2 ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr make dep make make install ldconfig -p | grep pj ldconfig wget http://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-13.8-current.tar.gz [root@siptest asterisk-certified-13.8-cert1]# ./configure --libdir=/usr/lib64 [root@siptest asterisk-certified-13.8-cert1]# make menuselect [root@siptest asterisk-certified-13.8-cert1]# make && make install Forget the option "--with-pjproject-bundled" I would say. Did not work for me on : CentOS release 6.8 (Final) Kind regards. On 12-08-16 17:22, Jonas Kellens wrote: Hello running into several problems when installing asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and 12). I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled First, I do not seem to have res_srtp module available, although all necessary libs are present on the system Second, I am not able to start Asterisk with following error : "/usr/sbin/asterisk: error while loading shared libraries: libpj.so.2: cannot open shared object file: No such file or directory" Help appreciated. Kind regards. On 12-08-16 16:58, Jonas Kellens wrote: On 12-08-16 16:38, Joshua Colp wrote: Jonas Kellens wrote: Question : I noticed I received an error when installing pjproject --with-external-srtp I do not seems to have the srtp capability. (However I can easily install with "yum install libsrtp-devel") Can this have anything to do with the no-audio-problems that I'm having ?? WebRTC requires SRTP and Asterisk has to be built with it enabled. It's okay if pjproject doesn't as we don't use their media layer. Do you have the res_srtp module in Asterisk? Hello Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest version Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version However, I am not able to select res_srtp module in menuselect. It says XXX res_srtp module Kind regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
