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
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
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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
OH , Thank you sir , it is done . thank you again. Best regards Amani Le mer. 11 mai 2016 à 16:02, Matt Broad <matt@supportedbusiness.com> a écrit :
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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
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Alex Pierry
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amani mansour
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Matt Broad