SAIL Community Supported PBX

which sail ver and debian should i using?
thanks

ok the call to the cell was a direct call also tried call to landline same thing no audio
my firewall router is sme 8--other than sme  nothing but brain dead switches
i did have sark working on a previous machine on the current network -- 2 years old jessie i think--box died
any way the verify check -- it has been a while but tcp dump if i knew what to look for
thanks

ps is jeff from selentra still around

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'

sorry -- note to self WEAR YOUR GLASSES

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

thanks that was it -- no zpt -- missed the correct password but i fixed that
thank you very much now i can play with tcpdump and look at it -- 1 packet worth a 1000 words
screen and tcpdump would be nice to add to the basic package
rob

ok i can manually configure both phones t26 and t21  -- after a factory reset of the phones discovery will find and correctly identify both phones after i adopt the phones a correct cfg file named by the correct mac address is generated but the phones will not provision.  what is the format of the provisioning message  after the initialmsg below -- i will see if i can find it with tcpdump

Mar  1 16:54:08 sarksvr responder: IN -------------------------------------------------------{
Mar  1 16:54:08 sarksvr responder:
Mar  1 16:54:08 sarksvr responder: Via: SIP/2.0/UDP 192.168.10.196:5059;branch=z9hG4bK1488326427
Mar  1 16:54:08 sarksvr responder: From: <sip:MAC00156534e02a@224.0.1.75>;tag=1488326427
Mar  1 16:54:08 sarksvr responder: To: <sip:MAC00156534e02a@224.0.1.75>
Mar  1 16:54:08 sarksvr responder: Call-ID: 1488326427@192.168.10.196
Mar  1 16:54:08 sarksvr responder: CSeq: 1 SUBSCRIBE
Mar  1 16:54:08 sarksvr responder: Contact: <sip:MAC00156534e02a@192.168.10.196:5059>
Mar  1 16:54:08 sarksvr responder: Max-Forwards: 70
Mar  1 16:54:08 sarksvr responder: User-Agent: Yealink SIP-T26P 6.73.0.40
Mar  1 16:54:08 sarksvr responder: Expires: 0
Mar  1 16:54:08 sarksvr responder: Event: ua-profile;profile-type="device";vendor="yealink";model="SIP-T26P";version="6.73.0.40"
Mar  1 16:54:08 sarksvr responder: Accept: application/url
Mar  1 16:54:08 sarksvr responder: Content-Length: 0
Mar  1 16:54:08 sarksvr responder: ----------------------------------------------------------}

sorry what i posted was it for syslog  it was complete and the  the codec's were in the provisioning file  --  all after what i posted  is cron logging messages
i cannot see anything pertaining to the phone in the log when i unplug it

ok the last posted was complete except the setting of the codec's which looked redundant -- when is the second phase suppose to show up -- i don't see the second part that you talked about -- how long after the segment i posted should the second part show up? i think that might be the problem.
could you post a copy of the second part so i now what to look for ? and is it in the syslog file? if not where? -- i can use tcpdump

thank you
rob

seems to be pretty close also noticed that sail see's it go offline --



Mar  1 16:54:08 sarksvr responder: IN -------------------------------------------------------{
Mar  1 16:54:08 sarksvr responder:
Mar  1 16:54:08 sarksvr responder: Via: SIP/2.0/UDP 192.168.10.196:5059;branch=z9hG4bK1488326427
Mar  1 16:54:08 sarksvr responder: From: <sip:MAC00156534e02a@224.0.1.75>;tag=1488326427
Mar  1 16:54:08 sarksvr responder: To: <sip:MAC00156534e02a@224.0.1.75>
Mar  1 16:54:08 sarksvr responder: Call-ID: 1488326427@192.168.10.196
Mar  1 16:54:08 sarksvr responder: CSeq: 1 SUBSCRIBE
Mar  1 16:54:08 sarksvr responder: Contact: <sip:MAC00156534e02a@192.168.10.196:5059>
Mar  1 16:54:08 sarksvr responder: Max-Forwards: 70
Mar  1 16:54:08 sarksvr responder: User-Agent: Yealink SIP-T26P 6.73.0.40
Mar  1 16:54:08 sarksvr responder: Expires: 0
Mar  1 16:54:08 sarksvr responder: Event: ua-profile;profile-type="device";vendor="yealink";model="SIP-T26P";version="6.73.0.40"
Mar  1 16:54:08 sarksvr responder: Accept: application/url
Mar  1 16:54:08 sarksvr responder: Content-Length: 0
Mar  1 16:54:08 sarksvr responder: ----------------------------------------------------------}

sail           5.0.0-20 

this is the head of the provisioning file shown by the mac address from the browser -- i can add the rest of the codec's if needed
#!version:1.0.0.1

##File header "#!version:1.0.0.1" can not be edited or deleted, and must be placed in the first line.##

account.1.enable = 1
account.1.label = 401
account.1.display_name = Ext401
account.1.auth_name = 401
account.1.password = nVBM7wyJ 
account.1.user_name =  401
account.1.sip_server_host = 192.168.10.198
account.1.outbound_proxy_enable = 1
account.1.outbound_host = 192.168.10.198
account.1.proxy_require = 192.168.10.198

#Enable or disable the phone to subscribe the register status; 0-Disabled (default), 1-Enabled;
account.1.subscribe_register = 1

#Enable or disable the phone to subscribe the message waiting indicator; 0-Disabled (default), 1-Enabled;
account.1.subscribe_mwi = 1

#Enable or disable the phone to subscribe to the voicemail through the message waiting indicator; 0-Disabled (default), 1-Enabled;
account.1.subscribe_mwi_to_vm = 1

voice_mail.number.1 = *50*

# Enable/Disable the codecs you want to use - default is law, G729, G722

account.1.codec.1.enable = 1
account.1.codec.1.payload_type = PCMU
account.1.codec.1.priority = 1
account.1.codec.1.rtpmap = 0

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