PJSIP syntax error exception when parsing non-ascii callerid on asterisk

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This issue originates from this asterisk issue:

https://issues.asterisk.org/jira/browse/ASTERISK-24664

==================

While investigating messages.log for my asterisk server; I found multiple
occurrences of this pjsip exception.

It happens when the sip caller id is non-ascii, like this (first one
appears in the log)
- ???????
- ????
- ???????

This call is rejected from pjsip. Asterisk produces an error type log
message as below:

[Jan  6 10:23:39] ERROR[5758] pjsip:    sip_transport. Error processing 789
bytes packet from UDP 45.115.149.5:5060 : PJSIP syntax error exception when
parsing '' hea
der on line 3 col 29:
INVITE sip:87640475343804107 at 22.73.22.39 SIP/2.0
Via: SIP/2.0/UDP 45.115.149.5:5060;branch=z9hG4bK01d3b09b6b21b00e
From: "???????" <sip:???????@45.115.149.5>;tag=6e1d71e43ae3ba8b
To: <sip:87640475343804107 at 22.73.22.39>
Call-ID: 043cf6ec0cae62d963616c6c0dc6dd2c at 45.115.149.5
CSeq: 1 INVITE
Contact: <sip:???????@45.115.149.5:5060>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK
Supported: timer
Max-Forwards: 70
User-Agent: SomeSwitch V2.1.2.4
Session-Expires: 600
Content-Length: 222
Content-Type: application/sdp

v=0
o=- 46616 46616 IN IP4 73.26.42.78
s=VOS3000
c=IN IP4 73.26.42.78
t=0 0
m=audio 6654 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


Does PJSIP support non-ascii caller id?

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