[asterisk-users] PJSIP Qualify

[Ian McMaster]

I am currently not using qualify, but it seems like a nice way to know if
the phones are online.  I attempted to set it up, but am running into a 404
on the subscription.

1.  From the manager, Action: PJSIPNotify (with an endpoint).  This caused
the following OPTIONS packet to be sent to the phone.

OPTIONS sip:64167f3a7955 at 192.168.2.206 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.22:5060
;rport;branch=z9hG4bKPj232f200b-90a0-4c2b-8171-164f9961175d
From: <sip:64167f3a7955 at 192.168.2.22
>;tag=100c883f-a844-4a7a-9ab3-085b16825b30
To: <sip:64167f3a7955 at 192.168.2.206>
Contact: <sip:64167f3a7955 at 192.168.2.22:5060>
Call-ID: 68897833-b4d3-4a75-9ad1-939b752d5a5e
CSeq: 47099 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 16.1.0
Content-Length:  0

2.  The phone acknowledges the OPTIONS

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.22:5060
;rport;branch=z9hG4bKPj232f200b-90a0-4c2b-8171-164f9961175d
From: <sip:64167f3a7955 at 192.168.2.22
>;tag=100c883f-a844-4a7a-9ab3-085b16825b30
To: "64167f3a7955" <sip:64167f3a7955 at 192.168.2.206>;tag=B9366B50-E4C6248D
CSeq: 47099 OPTIONS
Call-ID: 68897833-b4d3-4a75-9ad1-939b752d5a5e
Contact: <sip:64167f3a7955 at 192.168.2.206>
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Supported: replaces,100rel
User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848
Accept-Language: en
Accept:
application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Supported: 100rel,replaces,norefersub,sdp-anat
Content-Length: 0

3.  The phone sends the SUBSCRIBE

SUBSCRIBE sip:64167f3a7955 at 192.168.2.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.206;branch=z9hG4bK392bc0f56C717732
From: "64167f3a7955" <sip:64167f3a7955 at 192.168.2.22>;tag=A23DD37B-4BFEDFB8
To: <sip:64167f3a7955 at 192.168.2.22>
CSeq: 1 SUBSCRIBE
Call-ID: 40b12c267977075518c68280933a7955
Contact: <sip:64167f3a7955 at 192.168.2.206>
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
Event: message-summary
User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848
Accept-Language: en
Accept: application/simple-message-summary
Max-Forwards: 70
Expires: 3600
Content-Length: 0

4.  However, I am getting a 404 back.

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.2.205;rport=5060;received=192.168.2.205;branch=z9hG4bKd563e8de3F0F438D
Call-ID: adf2f90a0788a47d709d3b21dc39a92f
From: "sip:64167f39a92f" <sip:64167f39a92f at 192.168.2.22
>;tag=D55C2C08-654F3C47
To: <sip:64167f39a92f at 192.168.2.22>;tag=z9hG4bKd563e8de3F0F438D
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 16.1.0
Content-Length:  0

The phones are otherwise registered, and have working subscriptions to
hints (i.e. for activity on DAHDI channels, or parking lots).

What other configuration is required to get Qualify working?  Thank you in
advance.
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