[asterisk-users] Asterisk PJSIP Presence/Subscription Setup with Cisco and Grandstream Phones

Sat Aug 14 09:09:24 CDT 2021

On Sat, Aug 14, 2021 at 10:36 AM Reuben Farrelly <
reuben-asterisk-users at reub.net> wrote:

<snip>

> Logs show:
>
> Aug 14 22:54:41] ERROR[26606] res_pjsip.c: Could not create dialog with
> endpoint 1001. Invalid URI (PJSIP_EINVALIDURI)
> [Aug 14 22:54:41] WARNING[26606] res_pjsip_pubsub.c: Unable to create
> dialog for SIP subscription
> [Aug 14 22:54:41] WARNING[26606] res_pjsip_pubsub.c: Failed recreating
> '1001' subscription: Could not create subscription tree.
> [Aug 14 22:54:41] ERROR[26606] res_pjsip.c: Could not create dialog with
> endpoint 1002. Invalid URI (PJSIP_EINVALIDURI)
> [Aug 14 22:54:41] WARNING[26606] res_pjsip_pubsub.c: Unable to create
> dialog for SIP subscription
> [Aug 14 22:54:41] WARNING[26606] res_pjsip_pubsub.c: Failed recreating
> '1002' subscription: Could not create subscription tree.
> [Aug 14 22:54:41] ERROR[26606] res_pjsip.c: Could not create dialog with
> endpoint 1001. Invalid URI (PJSIP_EINVALIDURI)
> [Aug 14 22:54:41] WARNING[26606] res_pjsip_pubsub.c: Unable to create
> dialog for SIP subscription
>

Did these occur at startup? If so, it's because generally you can't
recreate the subscription to a TCP or TLS based device.

>
>
> Debugs show this:
>
> <--- Received SIP request (628 bytes) from
> TCP:[2403:5800:7700:6411::5]:5066 --->
> SUBSCRIBE sip:1003 at sip.reub.net SIP/2.0
> Via: SIP/2.0/TCP [2403:5800:7700:6411::5]:5066;branch=z9hG4bK-8a5f8a6c
> From: "Reuben Farrelly's Phone" <sip:1002 at sip.reub.net
> >;tag=96894202b140ddb
> To: <sip:1003 at sip.reub.net>
> Call-ID: 1eb5588-6f32a075 at 2403:5800:7700:6411::5
> CSeq: 26191 SUBSCRIBE
> Max-Forwards: 70
> Contact: "Reuben Farrelly's Phone"
> <sip:1002@[2403:5800:7700:6411::5]:5066;transport=tcp>
> Accept: multipart/related
> Accept: application/rlmi+xml
> Accept: application/dialog-info+xml
> Expires: 1800
> Event: dialog
> User-Agent: Cisco-CP-8845-3PCC/11.3.4
> Content-Length: 0
> Supported: replaces, sec-agree, eventlist
>
>
> <--- Transmitting SIP response (538 bytes) to
> TCP:[2403:5800:7700:6411::5]:5066 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/TCP
>
> [2403:5800:7700:6411::5]:5066;rport=5066;received=2403:5800:7700:6411::5;branch=z9hG4bK-8a5f8a6c
> Call-ID: 1eb5588-6f32a075 at 2403:5800:7700:6411::5
> From: "Reuben Farrelly's Phone" <sip:1002 at sip.reub.net
> >;tag=96894202b140ddb
> To: <sip:1003 at sip.reub.net>;tag=z9hG4bK-8a5f8a6c
> CSeq: 26191 SUBSCRIBE
> WWW-Authenticate: Digest
>
> realm="asterisk",nonce="x",opaque="54aa5d4217da2b11",algorithm=md5,qop="auth"
> Server: Asterisk PBX 18.6.0
> Content-Length:  0
>
>
> <--- Received SIP request (887 bytes) from
> TCP:[2403:5800:7700:6411::5]:5066 --->
> SUBSCRIBE sip:1003 at sip.reub.net SIP/2.0
> Via: SIP/2.0/TCP [2403:5800:7700:6411::5]:5066;branch=z9hG4bK-e18fdc9c
> From: "Reuben Farrelly's Phone" <sip:1002 at sip.reub.net
> >;tag=96894202b140ddb
> To: <sip:1003 at sip.reub.net>
> Call-ID: 1eb5588-6f32a075 at 2403:5800:7700:6411::5
> CSeq: 26192 SUBSCRIBE
> Max-Forwards: 70
> Authorization: Digest
> username="1002",realm="asterisk",nonce="x",uri="sip:1003 at sip.reub.net
> ",algorithm=MD5,response="x",opaque="54aa5d4217da2b11",qop=auth,nc=00000001,cnonce="80ff50e8"
> Contact: "Reuben Farrelly's Phone"
> <sip:1002@[2403:5800:7700:6411::5]:5066;transport=tcp>
> Accept: multipart/related
> Accept: application/rlmi+xml
> Accept: application/dialog-info+xml
> Expires: 1800
> Event: dialog
> User-Agent: Cisco-CP-8845-3PCC/11.3.4
> Content-Length: 0
> Supported: replaces, sec-agree, eventlist
>
>
> <--- Transmitting SIP response (637 bytes) to
> TCP:[2403:5800:7700:6411::5]:5066 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/TCP
>
> [2403:5800:7700:6411::5]:5066;rport=5066;received=2403:5800:7700:6411::5;branch=z9hG4bK-e18fdc9c
> Call-ID: 1eb5588-6f32a075 at 2403:5800:7700:6411::5
> From: "Reuben Farrelly's Phone" <sip:1002 at sip.reub.net
> >;tag=96894202b140ddb
> To: <sip:1003 at sip.reub.net>;tag=5e48d14e-efda-4f73-b1e1-7dcbbe12e7c4
> CSeq: 26192 SUBSCRIBE
> Expires: 1800
> Contact: <sip:[2403:5800:7700:6410::25]:5060;transport=TCP>
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
> CANCEL, UPDATE, PRACK, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub
> Server: Asterisk PBX 18.6.0
> Content-Length:  0
>
>
> <--- Transmitting SIP request (949 bytes) to
> TCP:[2403:5800:7700:6411::5]:5066 --->
> NOTIFY sip:1002@[2403:5800:7700:6411::5]:5066;transport=tcp SIP/2.0
> Via: SIP/2.0/TCP
>
> [2403:5800:7700:6410::25]:5060;rport;branch=z9hG4bKPj21e9c122-5302-4c39-bee5-9ea0d97fd29c;alias
> From: <sip:1003 at sip.reub.net>;tag=5e48d14e-efda-4f73-b1e1-7dcbbe12e7c4
> To: "Reuben Farrelly's Phone" <sip:1002 at sip.reub.net>;tag=96894202b140ddb
> Contact: <sip:[2403:5800:7700:6410::25]:5060;transport=TCP>
> Call-ID: 1eb5588-6f32a075 at 2403:5800:7700:6411::5
> CSeq: 8775 NOTIFY
> Event: dialog
> Subscription-State: active;expires=1799
> Allow-Events: message-summary, presence, dialog, refer
> Max-Forwards: 70
> User-Agent: Asterisk PBX 18.6.0
> Content-Type: application/dialog-info+xml
> Content-Length:   258
>
> <?xml version="1.0" encoding="UTF-8"?>
> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
> state="full" entity="sip:1003@
> [2403:5800:7700:6410::25]:5060;transport=TCP">
>  <dialog id="1003">
>   <state>terminated</state>
>  </dialog>
> </dialog-info>
>
> <--- Received SIP response (396 bytes) from
> TCP:[2403:5800:7700:6411::5]:5066 --->
> SIP/2.0 200 OK
> To: "Reuben Farrelly's Phone" <sip:1002 at sip.reub.net>;tag=96894202b140ddb
> From: <sip:1003 at sip.reub.net>;tag=5e48d14e-efda-4f73-b1e1-7dcbbe12e7c4
> Call-ID: 1eb5588-6f32a075 at 2403:5800:7700:6411::5
> CSeq: 8775 NOTIFY
> Via: SIP/2.0/TCP
>
> [2403:5800:7700:6410::25]:5060;branch=z9hG4bKPj21e9c122-5302-4c39-bee5-9ea0d97fd29c;alias
> Server: Cisco-CP-8845-3PCC/11.3.4
> Content-Length: 0
>
>
> The <state>terminated><state> line stands out as something I would not
> expect.
>

This appears to be a perfectly normal subscription. For the NOTIFY It means
that the device is not in use. The dialog-info+xml type isn't exactly for
presence, it's to show you calls with the device. We feed hint status
information in and construct a packet, as best we can, to represent the
hint status. You'd need to provide a full SIP trace showing not just the
initial subscription, but also when it should have changed to in use.

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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