[asterisk-users] help on ${RTPAUDIOQOS}

[Klaus Darilion]

Do you have canreinvite=no in sip.conf? Maybe the variable is only set 
if Asterisk is actually relaying RTP too.

regards
klaus

Asterisk User wrote:
> Hi All,
> 
> While reading about QoS, I came across ${RTPAUDIOQOS} and tried to use 
> it in my dialplan.
> I had 2 sip extensions 555 and 666 and I called from 555 to 666, but 
> unfortunately no value for ${RTPAUDIOQOS} appeared on Asterisk CLI.
> 
> Would you please let me know what is wrong with my dialplan and/or what 
> else should be done to get the value of ${RTPAUDIOQOS}?
> 
> Following is my dialplan context where my call landed....
> 
> [incoming_vpbx]
> exten => _x.,1,NoOp(A call has come)
> exten => _x.,n,Noop(============${RTPAUDIOQOS})
> 
> exten => _x.,n,Dial(SIP/666,30,m)
> exten => _x.,n,Hangup()
> exten => h,1,Noop(***************${RTPAUDIOQOS})
> 
> 
> And here is what appeared on CLI...
>     -- Executing [12 at incoming_vpbx:1] NoOp("SIP/555-b7a80948", "A call 
> has come") in new stack
>     -- Executing [12 at incoming_vpbx:2] NoOp("SIP/555-b7a80948", 
> "============") in new stack
>     -- Executing [12 at incoming_vpbx:3] Dial("SIP/555-b7a80948", 
> "SIP/666,30,m") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called 666
>     -- Started music on hold, class 'default', on SIP/555-b7a80948
>     -- SIP/666-089cb090 is ringing
>     -- SIP/666-089cb090 answered SIP/555-b7a80948
>     -- Stopped music on hold on SIP/555-b7a80948
>     -- Packet2Packet bridging SIP/555-b7a80948 and SIP/666-089cb090
>     -- Executing [h at incoming_vpbx:1] NoOp("SIP/555-b7a80948", 
> "***************") in new stack
> 
> 
> Thanking you...
> 
> ---Asterisk User
> 
> 
> 
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