John Hughes
2020-Mar-23 17:44 UTC
[asterisk-users] Attempting to get BLF working with linphone
So I've got a bit further with my project to get BLF working between asterisk and linphone. Initially asterisk was rejecting linphone's SUBSCRIBE messages because they didn't have an Accept: header. I've fixed that and now the initial SUBSCRIBE messages work and I see all my online contacts in green. But after a few minutes linphone attempts to renew the subscriptions and asterisk is not happy at all: <--- SIP read from UDP:10.27.128.3:5060 ---> SUBSCRIBE sip:jacques at 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:john at masked.masked.com>;tag=iGH81k5xf To: <sip:jacques at 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 at 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 at 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:john at masked.masked.com>;tag=iGH81k5xf To: <sip:jacques at 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:jacques at masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport From: <sip:john at masked.masked.com>;tag=c3Wvuu2XH To: sip:jacques at masked.masked.com CSeq: 20 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: <sip:john at 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 at 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:john at masked.masked.com>;tag=c3Wvuu2XH To: sip:jacques at 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:jacques at masked.masked.com SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport From: <sip:john at masked.masked.com>;tag=c3Wvuu2XH To: sip:jacques at 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 at 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:jacques at 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:john at masked.masked.com>;tag=c3Wvuu2XH To: sip:jacques at 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 <------------> [Mar 23 18:23:09] WARNING[2128]: chan_sip.c:4071 retrans_pkt: Retransmission timeout reached on transmission SQOclJgm4O for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response Why is asterisk giving an error 500? I can find no reason, there is nothing in any log. Why does asterisk think the error 500 is going to be acked? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200323/668d45ca/attachment.html>
Joshua C. Colp
2020-Mar-23 17:51 UTC
[asterisk-users] Attempting to get BLF working with linphone
On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com> wrote:> So I've got a bit further with my project to get BLF working between > asterisk and linphone. > > Initially asterisk was rejecting linphone's SUBSCRIBE messages because > they didn't have an Accept: header. I've fixed that and now the initial > SUBSCRIBE messages work and I see all my online contacts in green. > > But after a few minutes linphone attempts to renew the subscriptions and > asterisk is not happy at all: > > > <--- SIP read from UDP:10.27.128.3:5060 ---> > SUBSCRIBE sip:jacques at 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:john at masked.masked.com>;tag=iGH81k5xf > To: <sip:jacques at 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 at 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 at 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:john at masked.masked.com>;tag=iGH81k5xf > To: <sip:jacques at 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:jacques at masked.masked.com SIP/2.0 > Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.oxfLJBaRw;rport > From: <sip:john at masked.masked.com>;tag=c3Wvuu2XH > To: sip:jacques at masked.masked.com > CSeq: 20 SUBSCRIBE > Call-ID: SQOclJgm4O > Max-Forwards: 70 > Supported: replaces, outbound > Event: presence > Expires: 600 > Accept: application/pidf+xml > Contact: <sip:john at 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 at 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:john at masked.masked.com>;tag=c3Wvuu2XH > To: sip:jacques at 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:jacques at masked.masked.com SIP/2.0 > Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.RNv418~xv;rport > From: <sip:john at masked.masked.com>;tag=c3Wvuu2XH > To: sip:jacques at 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 at 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:jacques at 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:john at masked.masked.com>;tag=c3Wvuu2XH > To: sip:jacques at 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 > > > <------------> > > [Mar 23 18:23:09] WARNING[2128]: chan_sip.c:4071 retrans_pkt: > Retransmission timeout reached on transmission SQOclJgm4O for seqno 103 > (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > Packet timed out after 32000ms with no response > > > 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.> Why does asterisk think the error 500 is going to be acked? >It doesn't. The message is for something else, it refers to sequence number 103. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200323/6fe4f3ef/attachment.html>
John Hughes
2020-Mar-23 22:29 UTC
[asterisk-users] Attempting to get BLF working with linphone
On 23/03/2020 18:51, Joshua C. Colp wrote:> On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com > <mailto:john at 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. > > Why does asterisk think the error 500 is going to be acked? > > It doesn't. The message is for something else, it refers to sequence > number 103.Ok, thanks, that's clear and obvious. Now I have to go beat up(*) the linphone people. ((*) in the nicest possible way of course). -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200323/ae154641/attachment.html>