DIAL outbound and place on hold

[sinoee]

asterisk version is 11.16
blower is cli: the question is why after dial the dialplan imediatly play on hold music…why not to play the early media .Hope for answer…

-- Executing [s@macro-dialGateway:1] Set("SIP/8761026-001226f5", "CALLERID(num)=83160194") in new stack
-- Executing [s@macro-dialGateway:2] Dial("SIP/8761026-001226f5", "SIP/13012345678@IBAC51,60,FgL(18000000:61000)") in new stack
   > Limit Data for this call:
   > timelimit      = 18000000 ms (18000.000 s)
   > play_warning   = 61000 ms (61.000 s)
   > play_to_caller = yes
   > play_to_callee = no
   > warning_freq   = 0 ms (0.000 s)
   > start_sound    = 
   > warning_sound  = timeleft
   > end_sound      = 

== Using SIP RTP CoS mark 5
– Called SIP/13012345678@IBAC51
> 0x7fe0dc20eed0 – Probation passed - setting RTP source address to 192.168.122.72:48344
– Call on SIP/IBAC51-001226f6 placed on hold
– Started music on hold, class ‘default’, on SIP/8761026-001226f5
– SIP/IBAC51-001226f6 is making progress passing it to SIP/8761026-001226f5
> 0x7fe0dc20eed0 – Probation passed - setting RTP source address to 192.168.122.72:48344
> 0x7fe17da26ee0 – Probation passed - setting RTP source address to 211.166.195.74:4020
> 0x7fe0dc20eed0 – Probation passed - setting RTP source address to 192.168.122.72:48344
– Stopped music on hold on SIP/8761026-001226f5
== Spawn extension (macro-dialGateway, s, 3) exited non-zero on ‘SIP/8761026-001226f5’ in macro ‘dialGateway’
== Spawn extension (Gdfs, 13012345678, 5) exited non-zero on ‘SIP/8761026-001226f5’

the sip is:

[jcolp]

Your version of Asterisk is over 5 years old, so be aware that your problem may already be solved and as well people may not want to spend time assisting with such an old version.

To actually see if a remote party has put that leg on hold you would need to provide an actual full SIP trace using “sip set debug on”.