no audio after sip connection

[rrkelly]

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'

[sysadmin]

Dynamic features doesn't have anything to do with this I'm afraid.   

One-way or no audio is almost always caused by NAT traversal issues.   Before you do anything else, ensure your local router has all SIP ALG's or SIP "assists" turned off.   Routers, both expensive and cheap, invariably screw up SIP NAT traversal so make sure yours isn't messing with the packets. 

You don't say if this is a direct call to the cell or a diverted call.   In any event, check the following...

1.  Ensure you have set the correct public IP address for your PBX in SARK.   Pre V6 this was done in the Globals panel and you had to set it manually.  In V6, SARK figures it out for itself but you can override it (in the network panel) if you have a local network with more than one public IP and you want to control which will use SIP.

2.  On diverted and twinned calls the two SIP endpoints can get into a race condition where each is waiting for the other to speak first in order to establish symmetrical RTP.   You can usually avoid this by forcing Asterisk to take the initial call off hook before making the outbound dial.   To do that, set "Late Termination" in Globals to NO (V5) or toggle it OFF (V6)

3. Sometimes, when (2) doesn't work, you have to make yourself the first speaker.   To do this turn on call progress in the SIP trunk you are using to carry the call.

Usually one of these.   

As an aside, to fix the DYNAMIC features message, uncomment the automon feature in asterisk->sark_features_featuremap.conf (assuming you want to implement automon, if not, you can just ignore the message).

 

[rrkelly]

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

[sysadmin]

If you can do a regular internal call to another extension and it flows audio OK then the issue is outside of the LAN.    If you've checked and set all the values I gave you then you can run tcpdump to create a PCAP file and make a call.    Then you can see where the RTP is going by using wireshark to analyse the PCAP file. Something like this will do

Code Select Expand



tcpdump -i {name of your NIC} -s0 -w dumpall.pcap



Download the PCAP and show it to wireshark.