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
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