Hi Amani,

the DTMF will only be sent if you hit the dialplan extension which you created.

Assuming that the extension does the bridging, you would use something similar to the below:

originate user/1000<context> <exten>

replace <exten> with the extension you created.  This will originate a call to user 1000 and run the extension given.

thanks

Matt

On 11 May 2016 at 14:51, amani mansour <amani.mansour2@gmail.com> wrote:

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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