asterisk12 pjsip cannot support dial "+1234"

I want to user the number format: +1234
but the call always failed . if I changed the number for 1234, it works ok.
is it a bug?

Show us the dialplan and the cli output.

– Executing [+1234@dial_out:1] Set(“PJSIP/6789-00000000”, “DESTNUM=+1234”) in new stack
– Executing [+1234@dial_out:2] Dial(“PJSIP/6789-00000000”, “PJSIP/+1234,300,ogM(screen^CDRS)”) in new stack
[Feb 14 13:37:03] WARNING[9384][C-00000000]: app_dial.c:2425 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)

if dial number 1234 ,it works ok!

[simpletrans_udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

[1234]
type=endpoint
transport=simpletrans_udp
context=dial_out
disallow=all
allow=ulaw
direct_media=no
aors=1234
auth=1234

[1234]
type=aor
max_contacts=10
;contact=sip:192.168.250.230:5060

[1234]
type=auth
auth_type=userpass
password=111111
username=1234
or

[+1234]
type=endpoint
transport=simpletrans_udp
context=dial_out
disallow=all
allow=ulaw
direct_media=no
aors=+1234
auth=+1234

[+1234]
type=aor
max_contacts=10
;contact=sip:192.168.250.230:5060

[+1234]
type=auth
auth_type=userpass
password=111111
username=+1234

It is bad practice for SIP device names to be the same as extension numbers. This issue is with device names not extensions.

thanks david55 !

but I have more questions yet. why does it works well if no “+”? should I write different config ?
if that,how can I config when the number and username are the same ?
in chan_sip, defaultip can be used or host specifying the destination ip ,how to use this function in chan_pjsip?