2nd user in conference call drops after 30 seconds

Hi There,

I am having a problem that is limited to one extension/user. When I call one of my internal extensions then I conference in the problem extension (112). Extension 112 will drop from the call after 30 seconds. If I call extension 112 direct it does not happen. If I call extension 112 and then conference in other extensions it does not happen. It happens consistently if I call another internal extension first and then conference in 112.


  • Freepbx 14
  • PBX is hosted offsite (Vultr)
  • All extensions are using PJSIP with Direct Media set to No
  • All extensions are using Grandstream GXP 2160 phones
  • All phones have NAT traversal set to Keep-Alive
  • All extension are remote from the PBX and from different external networks.
  • SIP ALG is disabled on all routers. All routers are the same Unifi Security Gateways.

I am not seeing any timeouts in the Asterisk log files but I do see an entry for:
Channel PJSIP/112-0000210f left ‘native_rtp’ basic-bridge

Any help is appreciated. This is just a weird obscure issue that is only happening in a very specific order/scenario.


Turn on SIP Debug provide a call log.

A wireshark would be helpful too.

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