Any of those variables with dashes are invalid, you should look those up, those are params.


On Mon, Dec 26, 2016 at 11:49 AM Jeff Chua <jeff.chua.linux@gmail.com> wrote:
I'm configured freeswitch with gsmopen but can't seem to figure where

why I only get voice out from SIP-client to mobile-phone but not voice

from mobile to SIP client.



When I tried to call from mobile-phone to IVR extension 5000 and press

2 for the standard echo test (9196), I could hear the echo back. When

I tried to call 9195 for the five second delay echo test, I couldn't

anything.



I've add these settings in vars.xml as well.



  <X-PRE-PROCESS cmd="set" data="media_mix_inbound_outbound_codecs=true"/>

  <X-PRE-PROCESS cmd="set" data="inbound-late-negotiation=false"/>

  <X-PRE-PROCESS cmd="set" data="inbound-zrtp-passthru=false"/>

  <X-PRE-PROCESS cmd="set" data="disable-transcoding=false"/>

  <X-PRE-PROCESS cmd="set" data="inherit_codec=false"/>

  <X-PRE-PROCESS cmd="set" data="bypass_media=false"/>

  <X-PRE-PROCESS cmd="set" data="proxy_media=false"/>



What am I missing?



Thanks,

Jeff



_________________________________________________________________________

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