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

[Matt Broad]

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 at 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 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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>
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