Hi Mr Alex , But in dialplan/default.xml i configure a dtmf ,when a user 1000 call the extension 1001 we will get rtp event in wireshark . but with this one originate user/1000 &bridge(user/1001) i don't get the dtmf ? can you help me please ? Best regards Amani Le mer. 11 mai 2016 à 14:39, Alex Pierry <alex@teclan.com.br> a écrit :
originate user/1000<context> &bridge(user/1001@<context>)
------------------------------ *De:* freeswitch-users-bounces@lists.freeswitch.org < freeswitch-users-bounces@lists.freeswitch.org> em nome de amani mansour < amani.mansour2@gmail.com> *Enviado:* quarta-feira, 11 de maio de 2016 10:17 *Para:* FreeSWITCH Users Help *Assunto:* [Freeswitch-users] make a call between two sip phones using fs_cli ?
Hi all , Can i make a call between 2 sip phones from fs_cli interface ? if yes ,and it is using originate haw can i do it please ?
2 extensions 1000 and 1001 . Best regards Amani _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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