Re: [asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway
On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote: > When you have an "identify" object configured, you should just use > "ip" as > the "identify_by", But isn't ip the highest priory check in the default value of endpoint_identifier_order and by extension, wouldn't an endpoint without an "identify_by" implicitly be treated as "identify_by=ip"? So isn't my existing configuring already implicitly using "identify_by=ip"? Cheers, b. signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway
On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote: > > What version of Asterisk 13.11.1 I know, I could stand to upgrade. > and what's the value of the "identify_by" > parameter for the endpoint? It doesn't have one. I guess you are implying it should have one. > When you have an "identify" object configured, you should just use > "ip" as > the "identify_by", I see. OK. Will give that a try. Cheers, b. signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway
On Sat, Jan 26, 2019 at 10:56 AM Brian J. Murrell wrote: > I have a trunk set up for the DID from my provider: > > [my_provider] > type=registration > outbound_auth=my_provider > server_uri=sip:sip.example.com > client_uri=sip:my_usern...@sip.example.com > retry_interval=60 > > [my_provider] > type=auth > auth_type=userpass > password=123456 > username=my_username > > [my_provider] > type=aor > contact=sip:sip.example.com:5060 > > [my_provider] > type=endpoint > context=from-my_provider > disallow=all > allow=ulaw > outbound_auth=my_provider > aors=my_provider > > [my_provider] > type=identify > endpoint=my_provider > match=sip.example.com > > And it registers fine: > > > > > == > > mytrunk/sip:sip.example.com my_provider > Registered > > > And when it gets an INVITE from my provider (192.168.0.1): > > <--- Received SIP request (997 bytes) from UDP:192.168.0.1:5060 ---> > INVITE sip:1235551212@10.75.22.5:5060 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK06d035fd;rport > Max-Forwards: 70 > From: "Fred Flintstone" ;tag=as539f9476 > To: > Contact: > Call-ID: 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060 > CSeq: 102 INVITE > User-Agent: foobar > Date: Sat, 26 Jan 2019 17:40:00 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Remote-Party-ID: "Fred Flintstone" >;party=calling;privacy=off;screen=no > Content-Type: application/sdp > Content-Length: 295 > > [SDP redacted] > > It logs an error: > > [Jan 26 12:40:00] NOTICE[21775]: res_pjsip/pjsip_distributor.c:525 > log_failed_request: Request 'INVITE' from '"Fred Flintstone" < > sip:4565551212@192.168.0.1>' failed for '192.168.0.1:5060' (callid: > 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060) - No matching endpoint > found > > But then goes on to complete the call: > > <--- Transmitting SIP response (352 bytes) to UDP:192.168.0.1:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.0.1:5060 > ;rport=5060;received=192.168.0.1;branch=z9hG4bK06d035fd > Call-ID: 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060 > From: "Fred Flintstone" ;tag=as539f9476 > To: > CSeq: 102 INVITE > Server: Asterisk PBX 13.11.1 > Content-Length: 0 > > [ launch into dialplan ] > > So why the spurious error when it was able to complete the call? > What version of Asterisk and what's the value of the "identify_by" parameter for the endpoint? When you have an "identify" object configured, you should just use "ip" as the "identify_by", > > Cheers, > b. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- *George Joseph* Digium - A Sangoma Company | Software Developer | Software Engineering 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct/fax: +1 256 428 6012 Check us out at: https://digium.com ยท https://sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] INVITE from DID: No matching endpoint found but completes the call anyway
I have a trunk set up for the DID from my provider: [my_provider] type=registration outbound_auth=my_provider server_uri=sip:sip.example.com client_uri=sip:my_usern...@sip.example.com retry_interval=60 [my_provider] type=auth auth_type=userpass password=123456 username=my_username [my_provider] type=aor contact=sip:sip.example.com:5060 [my_provider] type=endpoint context=from-my_provider disallow=all allow=ulaw outbound_auth=my_provider aors=my_provider [my_provider] type=identify endpoint=my_provider match=sip.example.com And it registers fine: == mytrunk/sip:sip.example.com my_provider Registered And when it gets an INVITE from my provider (192.168.0.1): <--- Received SIP request (997 bytes) from UDP:192.168.0.1:5060 ---> INVITE sip:1235551212@10.75.22.5:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK06d035fd;rport Max-Forwards: 70 From: "Fred Flintstone" ;tag=as539f9476 To: Contact: Call-ID: 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060 CSeq: 102 INVITE User-Agent: foobar Date: Sat, 26 Jan 2019 17:40:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "Fred Flintstone" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 295 [SDP redacted] It logs an error: [Jan 26 12:40:00] NOTICE[21775]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"Fred Flintstone" ' failed for '192.168.0.1:5060' (callid: 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060) - No matching endpoint found But then goes on to complete the call: <--- Transmitting SIP response (352 bytes) to UDP:192.168.0.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.1:5060;rport=5060;received=192.168.0.1;branch=z9hG4bK06d035fd Call-ID: 3ef877dc4477d8ce4aae29965c5d0875@192.168.0.1:5060 From: "Fred Flintstone" ;tag=as539f9476 To: CSeq: 102 INVITE Server: Asterisk PBX 13.11.1 Content-Length: 0 [ launch into dialplan ] So why the spurious error when it was able to complete the call? Cheers, b. signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
