[asterisk-users] Attempting to get BLF working with linphone

[Joshua C. Colp]

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>