[Freeswitch-users] make a call between two sip phones using fs_cli ?

[amani mansour]

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 at teclan.com.br> a écrit :

> originate user/1000<context> &bridge(user/1001@<context>)
>
> ------------------------------
> *De:* freeswitch-users-bounces at lists.freeswitch.org <
> freeswitch-users-bounces at lists.freeswitch.org> em nome de amani mansour <
> amani.mansour2 at 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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