On Sat, Aug 14, 2021 at 10:36 AM Reuben Farrelly < [email protected]> 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:[email protected] SIP/2.0 > Via: SIP/2.0/TCP [2403:5800:7700:6411::5]:5066;branch=z9hG4bK-8a5f8a6c > From: "Reuben Farrelly's Phone" <sip:[email protected] > >;tag=96894202b140ddb > To: <sip:[email protected]> > Call-ID: 1eb5588-6f32a075@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@2403:5800:7700:6411::5 > From: "Reuben Farrelly's Phone" <sip:[email protected] > >;tag=96894202b140ddb > To: <sip:[email protected]>;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:[email protected] SIP/2.0 > Via: SIP/2.0/TCP [2403:5800:7700:6411::5]:5066;branch=z9hG4bK-e18fdc9c > From: "Reuben Farrelly's Phone" <sip:[email protected] > >;tag=96894202b140ddb > To: <sip:[email protected]> > Call-ID: 1eb5588-6f32a075@2403:5800:7700:6411::5 > CSeq: 26192 SUBSCRIBE > Max-Forwards: 70 > Authorization: Digest > username="1002",realm="asterisk",nonce="x",uri="sip:[email protected] > ",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@2403:5800:7700:6411::5 > From: "Reuben Farrelly's Phone" <sip:[email protected] > >;tag=96894202b140ddb > To: <sip:[email protected]>;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:[email protected]>;tag=5e48d14e-efda-4f73-b1e1-7dcbbe12e7c4 > To: "Reuben Farrelly's Phone" <sip:[email protected]>;tag=96894202b140ddb > Contact: <sip:[2403:5800:7700:6410::25]:5060;transport=TCP> > Call-ID: 1eb5588-6f32a075@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:[email protected]>;tag=96894202b140ddb > From: <sip:[email protected]>;tag=5e48d14e-efda-4f73-b1e1-7dcbbe12e7c4 > Call-ID: 1eb5588-6f32a075@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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