[asterisk-users] PJSIP Transfer not working

[CDR]

I tried to do what I with regular SIP to Transfer a call via 302
Redirect. In asterisk 12 we need  to add the Tech, or not, but in any
case, there is no transfer done. The call is closed.
Here is a trace. How do I do this?


[Jul  9 21:39:29] DEBUG[47716][C-00000002]: pbx.c:4869
pbx_extension_helper: Launching 'Transfer'
    -- Executing [17274428141 at redirect:30]
Transfer("PJSIP/Client.1.1.1.1-00000002",
"PJSIP/17274428141;rn=+18134029999;npdi at 1.1.1.1") in new stack
[Jul  9 21:39:29] DEBUG[47716][C-00000002]: pbx.c:4869
pbx_extension_helper: Launching 'Verbose'
    -- Executing [17274428141 at redirect:31]
Verbose("PJSIP/Client.1.1.1.1-00000002", "2,Transferred:
17274428141;rn=+18134029999;npdi at 1.1.1.1") in new stack
  == Transferred: 17274428141;rn=+18134029999;npdi at 1.1.1.1
    -- Auto fallthrough, channel 'PJSIP/Client.1.1.1.1-00000002'
status is 'UNKNOWN'
[Jul  9 21:39:29] DEBUG[47716][C-00000002]: channel.c:2597
ast_softhangup_nolock: Soft-Hanging (0x10) up channel
'PJSIP/Client.1.1.1.1-00000002'
[Jul  9 21:39:29] DEBUG[47716][C-00000002]: channel.c:2753 ast_hangup:
Hanging up channel 'PJSIP/Client.1.1.1.1-00000002'
[Jul  9 21:39:29] DEBUG[47716][C-00000002]: chan_pjsip.c:1578
hangup_cause2sip: AST hangup cause 0 (no match found in PJSIP)
<--- Transmitting SIP response (369 bytes) to UDP:1.1.1.1:49260 --->
SIP/2.0 603 Decline
v: SIP/2.0/UDP 1.1.1.1:49260;rport;received=1.1.1.1;branch=z9hG4bK-d8754z-22994e127365d474-1---d8754z-
i: MmFjNDM4NDc2NmFhZWNiYTU2MDQ1YmNjNGVmYmMyOTY
f: "9544447408" <sip:9544447408 at 8.26.191.189>;tag=82c82c1d
t: <sip:17274428141 at 8.26.191.189>;tag=09f3a67a-f457-46d1-8d16-243478ac3859
CSeq: 1 INVITE
Reason: Q.850;cause=0
l:  0