[asterisk-bugs] [Asterisk 0014644]: Asterisk should transform SIP 503 code to SIP 500
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Mar 12 14:18:34 CDT 2009
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=14644
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Reported By: ibc
Assigned To:
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Project: Asterisk
Issue ID: 14644
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.4.23
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-03-11 11:08 CDT
Last Modified: 2009-03-12 14:18 CDT
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Summary: Asterisk should transform SIP 503 code to SIP 500
Description:
Hi, in the following simple dialplan:
exten => _X.,1,Dial(SIP/trunk1/${EXTEN})
exten => _X.,n,Hangup
In case the trunk1 replies "SIP/2.0 503 Service Unavailable" Asterisk uses
the same SIP code to reply upstream. Asterisk shouldn't do it and MUST
convert that 503 into 500.
503 means that a client receiving it should try the same request against
an alternate server (got via DNS SRV and so).
This is "clearly" defined is RFC 3261:
-----------------------
21.5.4 503 Service Unavailable
[...]
A client (proxy or UAC) receiving a 503 (Service Unavailable) SHOULD
attempt to forward the request to an alternate server. It SHOULD NOT
forward any other requests to that server for the duration specified
in the Retry-After header field, if present.
-----------------------
Since Asterisk keep the 503 and replies it to the client, Asterisk breaks
the SIP failover mechanism, since it forces a client to contact an
alternate server when it's not needed at all (Asterisk is still alive and
working).
The correct behaviour is easy: When Asterisk receives a 503 from leg_B it
must convert it to 500 in leg_A.
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Relationships ID Summary
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related to 0014653 Never reply 503, use 500 instead (don't...
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(0101682) mmichelson (administrator) - 2009-03-12 14:18
http://bugs.digium.com/view.php?id=14644#c101682
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Hmm, I see several cases of Asterisk responding with a 503, but the two
cases you presented should have both resulted in Asterisk responding with a
500. There's obviously something subtle that I am missing while reading the
code.
In both cases, the Dial application should have hung up the incoming
channel when the 503 was received. With the patch I supplied, this should
have resulted in Asterisk sending a 500 to the calling side. As I said,
there must be something subtle I'm missing in the code. I'll set up a SIPp
server to respond to my INVITEs with a 503 so I can test this. Thanks!
Issue History
Date Modified Username Field Change
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2009-03-12 14:18 mmichelson Note Added: 0101682
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