SAIL Community Supported PBX

which sail ver and debian should i using?
thanks

ok i can dial out and connect to my cell  but i have no audio
i notice this line in the call progress out
  -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (__DYNAMIC_FEATURES=automon#clear#outpause#outresume)

but i get this later in the output log
ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/401-00000002.

could this be the problem and how do i fix it

complete call log below

== Using SIP RTP CoS mark 5
       > 0x7fb9c8020340 -- Strict RTP learning after remote address set to: 192.168.10.197:11786
    -- Executing [15079953174@internal:1] AGI("SIP/401-00000002", "sarkhpe,OutCos,15079953174,,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- <SIP/401-00000002>AGI Script sarkhpe completed, returning 0
    -- Executing [15079953174@401opencos:1] AGI("SIP/401-00000002", "sarkhpe,OutCluster,15079953174,,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- <SIP/401-00000002>AGI Script sarkhpe completed, returning 0
    -- Executing [15079953174@qrxvtmny:1] AGI("SIP/401-00000002", "sarkhpe,OutRoute,flow_out_bnd,,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (__DYNAMIC_FEATURES=automon#clear#outpause#outresume)
    -- AGI Script Executing Application: (Set) Options: (CALLERID(number)=401)
    -- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=14400)
    -- Channel will hangup at 2020-07-12 03:30:23.003 CDT.
    -- AGI Script Executing Application: (Dial) Options: (SIP/15079953174@78769303,,T)
  == Using SIP RTP CoS mark 5
    -- Called SIP/15079953174@78769303
       > 0x7fb9a0009830 -- Strict RTP learning after remote address set to: 23.29.23.42:33502
    -- SIP/78769303-00000003 is ringing
    -- SIP/78769303-00000003 is making progress passing it to SIP/401-00000002
       > 0x7fb9c8020340 -- Strict RTP switching to RTP target address 192.168.10.197:11786 as source
       > 0x7fb9c8020340 -- Strict RTP learning complete - Locking on source address 192.168.10.197:11786
    -- SIP/78769303-00000003 answered SIP/401-00000002
[2020-07-11 23:30:31] WARNING[1289][C-00000002]: features_config.c:1366 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/401-00000002.
[2020-07-11 23:30:31] WARNING[1296][C-00000002]: features_config.c:1366 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/78769303-00000003.
    -- Channel SIP/78769303-00000003 joined 'simple_bridge' basic-bridge <bda3e27e-6cad-4f3a-a10f-5902e4332edf>
[2020-07-11 23:30:31] WARNING[1289][C-00000002]: features_config.c:1366 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/401-00000002.
    -- Channel SIP/401-00000002 joined 'simple_bridge' basic-bridge <bda3e27e-6cad-4f3a-a10f-5902e4332edf>
    -- Channel SIP/78769303-00000003 left 'simple_bridge' basic-bridge <bda3e27e-6cad-4f3a-a10f-5902e4332edf>
    -- Channel SIP/401-00000002 left 'simple_bridge' basic-bridge <bda3e27e-6cad-4f3a-a10f-5902e4332edf>
    -- <SIP/401-00000002>AGI Script sarkhpe completed, returning 0
    -- Auto fallthrough, channel 'SIP/401-00000002' status is 'ANSWER'

how do i add my sip provider from with in sark 6 dedian is there a howto for ver 6 debian
i know how to add it manually to sip.conf

i suspect this has to do with the previous topic -- i have the embedded debian sark ver 5 i have two yealink phones a t21 that is found and provisioned just fine
and a t26 that is found but will not provision sark says #include yealink12 in the provisioning field for the t26 tried to change to t 26 no diff-- i tried to find config fields in the t26 phone itself to match what the previous topic mentioned still can not get the t26 to work -- the firmware for t26 is current any idea's is there a new t26 file?
rob