[asterisk-bugs] [JIRA] (ASTERISK-28299) 481 Call/Transaction Does Not Exist on receiving MESSAGE event
Brian J. Murrell (JIRA)
noreply at issues.asterisk.org
Thu Mar 7 11:44:47 CST 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-28299?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=246477#comment-246477 ]
Brian J. Murrell commented on ASTERISK-28299:
---------------------------------------------
{noformat}
<--- Received SIP request (1003 bytes) from UDP:192.168.5.6:5060 --->
INVITE sip:5555551212 at 10.75.22.8:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.6:5060;branch=z9hG4bK7539c954;rport
Max-Forwards: 70
From: "Fred Flintsone" <sip:5555551212 at 192.168.5.6>;tag=as6fe86629
To: <sip:5555551212 at 10.75.22.8:5060>
Contact: <sip:5555551212 at 192.168.5.6:5060>
Call-ID: 644970b50c7200aa1f7cc40d5f925346 at 192.168.5.6:5060
CSeq: 102 INVITE
User-Agent: itsp.example.com
Date: Wed, 06 Mar 2019 20:24:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Fred Flintsone" <sip:5555551212 at 192.168.5.6>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 297
v=0
o=root 1784943158 1784943158 IN IP4 192.168.5.6
s=itsp.example.com
c=IN IP4 192.168.5.6
t=0 0
m=audio 11350 RTP/AVP 0 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<--- Transmitting SIP response (526 bytes) to UDP:192.168.5.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.5.6:5060;rport=5060;received=192.168.5.6;branch=z9hG4bK7539c954
Call-ID: 644970b50c7200aa1f7cc40d5f925346 at 192.168.5.6:5060
From: "Fred Flintsone" <sip:5555551212 at 192.168.5.6>;tag=as6fe86629
To: <sip:5555551212 at 10.75.22.8>;tag=z9hG4bK7539c954
CSeq: 102 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1551903885/259e10745dba6fcf8820b48fdcbd27c0",opaque="4e82c5c17a353b76",algorithm=md5,qop="auth"
Server: Asterisk PBX 13.25.0
Content-Length: 0
<--- Received SIP request (426 bytes) from UDP:192.168.5.6:5060 --->
ACK sip:5555551212 at 10.75.22.8:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.6:5060;branch=z9hG4bK7539c954;rport
Max-Forwards: 70
From: "Fred Flintsone" <sip:5555551212 at 192.168.5.6>;tag=as6fe86629
To: <sip:5555551212 at 10.75.22.8:5060>;tag=z9hG4bK7539c954
Contact: <sip:5555551212 at 192.168.5.6:5060>
Call-ID: 644970b50c7200aa1f7cc40d5f925346 at 192.168.5.6:5060
CSeq: 102 ACK
User-Agent: itsp.example.com
Content-Length: 0
<--- Received SIP request (1286 bytes) from UDP:192.168.5.6:5060 --->
INVITE sip:5555551212 at 10.75.22.8:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.6:5060;branch=z9hG4bK579e73f7;rport
Max-Forwards: 70
From: "Fred Flintsone" <sip:5555551212 at 192.168.5.6>;tag=as6fe86629
To: <sip:5555551212 at 10.75.22.8:5060>
Contact: <sip:5555551212 at 192.168.5.6:5060>
Call-ID: 644970b50c7200aa1f7cc40d5f925346 at 192.168.5.6:5060
CSeq: 103 INVITE
User-Agent: itsp.example.com
Authorization: Digest username="[redacted]", realm="asterisk", algorithm=MD5, uri="sip:5555551212 at 172.0.0.1:5060", nonce="1551903885/259e10745dba6fcf8820b48fdcbd27c0", response="9623d76a01682c22ba375aabf5d746c0", opaque="4e82c5c17a353b76", qop=auth, cnonce="6a43566c", nc=00000001
Date: Wed, 06 Mar 2019 20:24:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Fred Flintsone" <sip:5555551212 at 192.168.5.6>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 297
v=0
o=root 1784943158 1784943159 IN IP4 192.168.5.6
s=itsp.example.com
c=IN IP4 192.168.5.6
t=0 0
m=audio 11350 RTP/AVP 0 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<--- Transmitting SIP response (354 bytes) to UDP:192.168.5.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.6:5060;rport=5060;received=192.168.5.6;branch=z9hG4bK579e73f7
Call-ID: 644970b50c7200aa1f7cc40d5f925346 at 192.168.5.6:5060
From: "Fred Flintsone" <sip:5555551212 at 192.168.5.6>;tag=as6fe86629
To: <sip:5555551212 at 10.75.22.8>
CSeq: 103 INVITE
Server: Asterisk PBX 13.25.0
Content-Length: 0
{noformat}
> 481 Call/Transaction Does Not Exist on receiving MESSAGE event
> --------------------------------------------------------------
>
> Key: ASTERISK-28299
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-28299
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Resources/res_pjsip_messaging
> Affects Versions: 13.25.0
> Reporter: Brian J. Murrell
> Assignee: Brian J. Murrell
> Severity: Minor
>
> When my ITSP tries to send a (SIP SIMPLE) MESSAGE event to my Asterisk machine, Asterisk responds with {{SIP/2.0 481 Call/Transaction Does Not Exist}} which doesn't really make any sense to me for a single/one-off event like a MESSAGE event. MESSAGE events don't seem to involve "transactions" which must exist.
> My ITSP configuration:
> {noformat}
> [trunk]
> type=registration
> transport=transport-udp
> outbound_auth=trunk-auth
> server_uri=sip:itsp.example.com
> client_uri=sip:userid at itsp.example.com
> [trunk-auth]
> type=auth
> auth_type=userpass
> password=********
> username=userid
> [trunk-endpoint](!)
> type=endpoint
> transport=transport-udp
> context=from-trunk
> message_context=messages
> disallow=all
> allow=ulaw
> from_user=userid
> outbound_auth=trunk-auth
> auth=trunk-auth
> send_pai=yes
> [trunk-aor](!)
> type=aor
> qualify_frequency=15
> [trunk-foo](trunk-endpoint)
> aors=trunk-foo
> [trunk-foo](trunk-aor)
> contact=sip:userid at itsp.example.com:5060
> [trunk-foo]
> type=identify
> endpoint=trunk-foo
> match=itsp.example.com
> {noformat}
> The SIP conversation when the ITSP is trying to send the MESSAGE:
> {noformat}
> <--- Received SIP request (456 bytes) from UDP:10.0.0.1:5060 --->
> MESSAGE sip:s at 10.75.22.8:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK51d6d4da;rport
> Max-Forwards: 70
> From: "5555551212" <sip:5555551212 at itsp.example.com>;tag=as6c34cb69
> To: <sip:s at 10.75.22.8:5060>
> Contact: <sip:5555551212 at 10.0.0.1:5060>
> Call-ID: 3e9735b4313e58f90b4b61c82f392c2e at 10.0.0.1:5060
> CSeq: 102 MESSAGE
> User-Agent: itsp.example.com
> X-SMS-To: 5551234567
> Content-Type: text/plain;charset=UTF-8
> Content-Length: 1
> test message
> <--- Transmitting SIP response (515 bytes) to UDP:10.0.0.1:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 10.0.0.1:5060;rport=5060;received=10.0.0.1;branch=z9hG4bK51d6d4da
> Call-ID: 3e9735b4313e58f90b4b61c82f392c2e at 10.0.0.1:5060
> From: "5555551212" <sip:5555551212 at itsp.example.com>;tag=as6c34cb69
> To: <sip:s at 10.75.22.8>;tag=z9hG4bK51d6d4da
> CSeq: 102 MESSAGE
> WWW-Authenticate: Digest realm="asterisk",nonce="1550441754/[redacted]",opaque="2c504a1035f74a1d",algorithm=md5,qop="auth"
> Server: Asterisk PBX 13.25.0
> Content-Length: 0
> <--- Received SIP request (707 bytes) from UDP:10.0.0.1:5060 --->
> MESSAGE sip:s at 10.75.22.8:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK516e2ba7;rport
> Max-Forwards: 70
> From: "5555551212" <sip:5555551212 at itsp.example.com>;tag=as6c34cb69
> To: <sip:s at 10.75.22.8:5060>;tag=z9hG4bK51d6d4da
> Call-ID: 3e9735b4313e58f90b4b61c82f392c2e at 10.0.0.1:5060
> CSeq: 103 MESSAGE
> User-Agent: itsp.example.com
> Proxy-Authorization: Digest username="userid", realm="asterisk", algorithm=MD5, uri="sip:s at 172.1.2.3:5060", nonce="1550441754/[redacted]", response="[redacted]", opaque="2c504a1035f74a1d", qop=auth, cnonce="544772bb", nc=00000002
> X-SMS-To: 5551234567
> Content-Type: text/plain;charset=UTF-8
> Content-Length: 1
> c
> <--- Transmitting SIP response (388 bytes) to UDP:10.0.0.1:5060 --->
> SIP/2.0 481 Call/Transaction Does Not Exist
> Via: SIP/2.0/UDP 10.0.0.1:5060;rport=5060;received=10.0.0.1;branch=z9hG4bK516e2ba7
> Call-ID: 3e9735b4313e58f90b4b61c82f392c2e at 10.0.0.1:5060
> From: "5555551212" <sip:5555551212 at itsp.example.com>;tag=as6c34cb69
> To: <sip:s at 10.75.22.8>;tag=z9hG4bK51d6d4da
> CSeq: 103 MESSAGE
> Server: Asterisk PBX 13.25.0
> Content-Length: 0
> {noformat}
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