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 defaultnameR4
defaultuserR4
secret=
dtmfmode=RFC2833
host=dynamic context=sip-exten qualify=yes rtptimeout`
rtpholdtimeout`
rtpkeepalive`
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@X
Via: SIP/2.0/UDP 3X:5060;branch=z9hG4bK2795cec0;rport Max-Forwards: 70
From: “WIRELESS CALLER” ;tag=as45ffbb22
To:
Contact:
Call-ID: 4f83ccef5bfaebf55271bc674e26165d@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

–