Hello Ruddy, Step 1: install FreeSwitch $ sudo yum install git autoconf automake libtool ncurses-devel libjpeg-devel $ sudo yum install expat-devel openssl-devel libtiff-devel libX11-devel unixODBC-devel libssl-devel python-devel \ zlib-devel libzrtpcpp-devel alsa-lib-devel libogg-devel libvorbis-devel perl-libs gdbm-devel \ libdb-devel uuid-devel @development-tools $ git clone git://git.freeswitch.org/freeswitch.git $ ./bootstrap.sh $ ./configure $ make $ make install $ make all install cd-sounds-install cd-moh-install $ /usr/local/freeswitch/bin/freeswitch Step 2: after installing the FS find a default username/password from: $ grep password /usr/local/freeswitch/conf/directory/default/1000.xml $ grep password /usr/local/freeswitch/conf/vars.xml Step 3: Follow up this http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-December/016904.html where installation is shown for pjsip Step 4: make call from pjsua by registerting to FreeSwitch see commands how to use there Report back community about the: audio playback and audio capture status registration and all the rest will work. Thanks Regards On Wed, Dec 4, 2013 at 7:46 AM, Ruddy Gbaguidi <plugworld at micnes.com> wrote: > Hi All, > > I have been looking for example on how to use PJSIP to create a PBX SIP > server which will have the features : > > - Accept registrations and send response back > > - Accept invite from a SIP endpoint, answer the call and play a > local file > > - Bridge call between two registered endpoints. > > > > Do you know what modules I should have or how it can be done ? > > > > Thanks > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20131204/af44396e/attachment-0001.html>