No voice in transfered call( no RTP )

[hsunryou]

my asterisk is Asterisk 16 + CentOS 6(pjsip. not chan_sip.so)

Inbound

GATEWAY —> Asterisk 16[Public IP] ----> IP-Phone(9001)
it works well with RTP.

But When i transfer call at dial-plan like this
[Transfer]
exten => _XXXXX.,1,Noop(–Transfer–)
same => n,Noop(CALLER={CALLERID(num)}) same => n,Noop(INEXTEN={INEXTEN})
same => n,Dial(PJSIP/${EXTEN}@GATEWAY,30,tTwWR)
same => n,Hangup

Call is connected. but here’s no voice.
so I check the RTP with “rtp set debug on” command and tcpdump
there’s no rtp packet!

how can i set the asterisk for transfering call?

[david551]

All I see here is normal dialplan for a call via the provider; I don’t see a transfer.

You would need to provide the SDP exchanges to fully understand this, but it is likely that you have a problem with trying to use direct media when the provider doesn’t support it.

In future, please use </> on the forum’s formatting menu, so that dialplan (and logs) are displayed correctly.