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@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
To: sip:jacq...@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@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
<------------>
[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?
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