[asterisk-bugs] [Asterisk 0018736]: [patch] Out-of-dialog MWI from a SIP Trunk in asterisk 1.8.2.2 always responds with 489 Bad event
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Feb 3 12:10:56 CST 2011
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18736
======================================================================
Reported By: cgoudie
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 18736
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.8.2.3
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2011-02-02 15:19 CST
Last Modified: 2011-02-03 12:10 CST
======================================================================
Summary: [patch] Out-of-dialog MWI from a SIP Trunk in
asterisk 1.8.2.2 always responds with 489 Bad event
Description:
Issue 13028 is said to have fixed this issue, however in 1.8 when I send an
out-of-dialog notify, I only receive 489 Bad event
Here is an example of the sip notify and response from wireshark: (.23 is
the Asterisk server. .47 is the host sending the sip notify)
---- From host sending sip notify ----
NOTIFY sip:5820 at 10.16.0.23 SIP/2.0
From:
"Anonymous"<sip:Anonymous at 10.16.0.47:5065>;tag=beeafd8-0-13c9-50022-ffdc7-52327663-ffdc7
To: <sip:5820 at 10.16.0.23:5060>
Call-ID: beeafd8-0-13c9-50022-ffdc7-4db95f02-ffdc7
CSeq: 1 NOTIFY
Via: SIP/2.0/UDP 10.16.0.47:5065;branch=z9hG4bK-ffdc7-3e7753f9-3403d179
Max-Forwards: 70
Supported: replaces
Contact: <sip:Anonymous at 10.16.0.47>
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 47
Messages-Waiting:yes
Voice Message: 1/0 (0/0).
---- Response from Asterisk server ----
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP
10.16.0.47:5065;branch=z9hG4bK-ffdc7-3e7753f9-3403d179;received=10.16.0.47
From:
"Anonymous"<sip:Anonymous at 10.16.0.47:5065>;tag=beeafd8-0-13c9-50022-ffdc7-52327663-ffdc7
To: <sip:5820 at 10.16.0.23:5060>;tag=as56d33dc9
Call-ID: beeafd8-0-13c9-50022-ffdc7-4db95f02-ffdc7
CSeq: 1 NOTIFY
Server: FPBX-2.8.1(1.8.2.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
--------
I have a SIP phone (grandstream) on extension 5820, and if I leave a voice
mail message with asterisk for that phone, it does activate the MWI light.
In doing some additional debugging, I see the code path headed through
(around line 20539) if(!p->mwi). it then searches for a peer on around
20541 and doesn't find one.
Additionally, the line char *c = ast_strdupa(get_body(req,
"Voice-Message", ':')); doesn't appear to place any text into c as a
result of these two packets.
My sip trunk's inbound is context=from-internal and it's able to dial the
extension 5820 fine. My phone's mailbox is 5820 at default
We are sending these out of dialog notifies from a SIP
auto-attendant/voicemail/ivr system. I can change how these packets look
(to an extent), however it's worth noting these packets currently work with
cisco call manager.
======================================================================
----------------------------------------------------------------------
(0131462) cmaj (reporter) - 2011-02-03 12:10
https://issues.asterisk.org/view.php?id=18736#c131462
----------------------------------------------------------------------
cgoudie, MS claims its MWI is RFC 3842 compliant in
http://technet.microsoft.com/en-us/library/dd298001.aspx but double
checking http://www.ietf.org/rfc/rfc3842.txt states in Section 3.5:
If the Request-URI or To header in a message-summary subscription
corresponds to a group or collection of individual messaging
accounts, the notifier MUST specify to which account the message-
summary body corresponds. Note that the account URI MUST NOT be
delimited with angle brackets ("<" and ">").
Message-Account: sip:alice at example.com
...so I think we need to take "Message-Account" & "To"/"Request-URI" into
account, perhaps looking for "Message-Account" first as my patch does AND
adding in "To" (and "Request-URI") parsing as you suggest, especially in
the case where "Message-Account" was not set. Provided it exists, however,
it looks like "Message-Account" will always be of finer accuracy than "To".
Issue History
Date Modified Username Field Change
======================================================================
2011-02-03 12:10 cmaj Note Added: 0131462
======================================================================
More information about the asterisk-bugs
mailing list