[asterisk-users] No remote address on RTP instance - On Ringing

[Nick Cameo]

Yes of course, I just did not want to overwhelm you guys with SIP
trace. Before that though, I realized something:

[Sep 10 12:03:30] WARNING[8178]: res_musiconhold.c:802 set_moh_exec:
SetMusicOnHold application is deprecated and will be removed. Use
Set(CHANNEL(musicclass)=...) instead
 -- AGI Script Executing Application: (DIAL) Options:
(SIP/VTrunk/19042572451,60,HRrL(240000:61000:30000)m)

There is that `m` option that jg was referring to. However, in
a2billing, I have made sure there is no `m` option in the `Dial
Command Params`: ,60,HRrL(%timeout%:61000:30000). The extension for
the entry does not include the option either:

exten => 1000,1,Answer
exten => 1000,n,Wait(1)
exten => 1000,n,AGI(a2billing.php)
exten => 1000,n,Wait(1)
exten => 1000,n,Hangup

Will run a test call with trace right now.

Kind Regards,

Nick.