[asterisk-users] POlycom phone not ringing behind firewall (401 permission denied)

[Jerry Geis]

Hi All,

I have polycom phones setup in an office connected to a cloud asterisk
server.
The polycom phones can call out just fine - audio just fine.
However a call coming into the cloud asterisk answers fine - get the
autoattendant, enter the extension and the polycom does not ring. The CLI
shows that the correct SIP extension is being Dialed  (SIP/524)

Looks like I'm getting a 401 permission denied.

What might I be missing here ?

my 524 extensions has:
[524]
type=friend
defaultname=524
defaultuser=524
secret=<yes>
dtmfmode=RFC2833
host=dynamic
context=sip-exten
qualify=yes
rtptimeout=60
rtpholdtimeout=60
rtpkeepalive=60
callerid="524 524" <524>
qualify=no
canreinvite=yes
timezone=1
nat=force_rport,comedia
disallow=all
allow=g722
allow=ulaw
allow=alaw
allow=gsm

Thanks,

Jerry

  == Using SIP RTP CoS mark 5
Audio is at 15876
Adding codec ulaw to SDP
Adding codec g722 to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 1X
INVITE sip:524 at X
Via: SIP/2.0/UDP 3X:5060;branch=z9hG4bK2795cec0;rport
Max-Forwards: 70
From: "WIRELESS CALLER" <sip:X>;tag=as45ffbb22
To: <sip:524 at X>
Contact: <sip:X:5060>
Call-ID: 4f83ccef5bfaebf55271bc674e26165d at X:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.33.0
Date: Tue, 30 Jun 2020 12:17:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 422927332 422927332 IN IP4 X
s=Asterisk PBX 13.33.0
c=IN IP4 X
t=0 0
m=audio 15876 RTP/AVP 0 9 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
    -- Called SIP/524
Retransmitting #6 (no NAT) to X
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP X:56790;branch=z9hG4bK334206641;received=X
From: <sip:201 at X>;tag=1340133334
To: <sip:X at X>;tag=as181c1453
Call-ID: 192924635-1732461672-2061354149
CSeq: 1 INVITE
Server: Asterisk PBX 13.33.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2625e522"
Content-Length: 0
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