Hi John, 1. Could you get any further, in your quest for working BLF with linphone ? 2. Have you tried with a different Linphone version (4.12 is pending on Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ?
Best regards Le mer. 25 mars 2020 à 15:06, John Hughes <j...@calva.com> a écrit : > > On 23/03/2020 18:51, Joshua C. Colp wrote: > > On Mon, Mar 23, 2020 at 2:45 PM John Hughes <j...@calva.com> wrote: > >> >> Why is asterisk giving an error 500? I can find no reason, there is >> nothing in any log. >> > > The sequence number is from the past. The first SUBSCRIBE is sequence > number 22 (check the CSeq header). The second is 20. The third is 21. It > appears as though this is from the past, so it receives a 500. > > Ok, I've had some back and forth with the linphone developers and they > contend that although the sequence number on the 2nd and 3rd SUBSCRIBE > messages start a new sequence this is legal as it is a new conversation -- > the "tag=" on the From has changed. > > Are they right? (Notice that the tag= from asterisk also changes). > > <--- SIP read from UDP:10.27.128.3:5060 ---> > SUBSCRIBE sip:jacques@10.27.128.1:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;rport > From: <sip:j...@masked.masked.com>;*tag=iGH81k5xf* > To: <sip:jacq...@masked.masked.com>;tag=as3c7de68c > CSeq: 22 SUBSCRIBE > Call-ID: SQOclJgm4O > Max-Forwards: 70 > Supported: replaces, outbound > Event: presence > Expires: 600 > Accept: application/pidf+xml > Contact: <sip:john@10.27.128.3;transport=udp> > ;+sip.instance="<urn:uuid:abcdf51a-82e0-49b9-a8ab-2461011f25ec>" > User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) > Authorization: Digest realm="asterisk", nonce="188b095b", algorithm=MD5, > username="john", uri="sip:jacques@10.27.128.1:5060", > response="bdbc7cbac4453fd643050bf28996a68e" > > <-------------> > --- (14 headers 0 lines) --- > Found peer 'john' for 'john' from 10.27.128.3:5060 > > <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 10.27.128.3:5060 > ;branch=z9hG4bK.NYP-ux0Zx;received=10.27.128.3;rport=5060 > From: <sip:j...@masked.masked.com>;*tag=iGH81k5xf* > To: <sip:jacq...@masked.masked.com>;tag=as3c7de68c > Call-ID: SQOclJgm4O > CSeq: 22 SUBSCRIBE > Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > nonce="3144c0a9", stale=true > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: > SUBSCRIBE) > > <--- SIP read from UDP:10.27.128.3:5060 ---> > SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 > Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport > From: <sip:j...@masked.masked.com>;*tag=c3Wvuu2XH <===== new > conversation* > To: sip:jacq...@masked.masked.com > CSeq: *20 SUBSCRIBE <=== sequence restarts* > Call-ID: SQOclJgm4O > Max-Forwards: 70 > Supported: replaces, outbound > Event: presence > Expires: 600 > Accept: application/pidf+xml > Contact: <sip:john@10.27.128.3;transport=udp> > ;+sip.instance="<urn:uuid:abcdf51a-82e0-49b9-a8ab-2461011f25ec>" > User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) > > <-------------> > --- (13 headers 0 lines) --- > Sending to 10.27.128.3:5060 (no NAT) > Creating new subscription > Sending to 10.27.128.3:5060 (no NAT) > sip_route_dump: route/path hop: <sip:john@10.27.128.3;transport=udp> > Found peer 'john' for 'john' from 10.27.128.3:5060 > > <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 10.27.128.3:5060 > ;branch=z9hG4bK.oxfLJBaRw;received=10.27.128.3;rport=5060 > From: <sip:j...@masked.masked.com>;tag=c3Wvuu2XH > To: sip:jacq...@masked.masked.com;tag=as007ffc64 > Call-ID: SQOclJgm4O > CSeq: 20 SUBSCRIBE > Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4224acfb" > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog 'SQOclJgm4O' in 32000 ms (Method: > SUBSCRIBE) > > <--- SIP read from UDP:10.27.128.3:5060 ---> > SUBSCRIBE sip:jacq...@masked.masked.com SIP/2.0 > Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport > From: <sip:j...@masked.masked.com>;tag=c3Wvuu2XH > To: sip:jacq...@masked.masked.com > CSeq: 21 SUBSCRIBE > Call-ID: SQOclJgm4O > Max-Forwards: 70 > Supported: replaces, outbound > Event: presence > Expires: 600 > Accept: application/pidf+xml > Contact: <sip:john@10.27.128.3;transport=udp> > ;+sip.instance="<urn:uuid:abcdf51a-82e0-49b9-a8ab-2461011f25ec>" > User-Agent: Linphone/3.12.0 (belle-sip/1.6.3) > Authorization: Digest realm="asterisk", nonce="4224acfb", algorithm=MD5, > username="john", uri="sip:jacq...@masked.masked.com", > response="eb30a9801e78d2cb2c58c61200c50cb1" > > <-------------> > --- (14 headers 0 lines) --- > > <--- Transmitting (no NAT) to 10.27.128.3:5060 ---> > *SIP/2.0 500 Server error* > Via: SIP/2.0/UDP 10.27.128.3:5060 > ;branch=z9hG4bK.RNv418~xv;received=10.27.128.3;rport=5060 > From: <sip:j...@masked.masked.com>;tag=c3Wvuu2XH > To: sip:jacq...@masked.masked.com;tag=as3c7de68c > Call-ID: SQOclJgm4O > CSeq: 21 SUBSCRIBE > Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Length: 0 > > > <------------> > -- > _____________________________________________________________________ > -- 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
-- _____________________________________________________________________ -- 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