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kill-the-user business ;-)
Why do we suddenly have two flags to set peer type?
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can close
some bug reports.
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supporting devices. The devices (snoms, specifically) need to receive a SIP
URI instead of just an extension. This adds that functionality.
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* Added doxygen comments to the major dahdi structures.
* Fixed PRI and SS7 using an incorrect string value if the extension
delimiter is not present in the Dial() function.
* Fixed SS7 not checking if the dialed extension is at least as long
as the stripmsd option.
* Fixed PRI not handling unknown TON/NPI prefix letters correctly.
* Fixed some uninitialized string variables on FXS ports.
configs/chan_dahdi.conf.sample
* Updated some documentation.
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Also, implement a private cause code (as suggested by Tilghman). This works with
chan_sip, but doesn't propagate through chan_local.
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calls.
- Also, change a function in app.c to return a userful value instead of always returning 0.
Patch by fnordian, changed by Corydon76 and myself.
This does not close the bug report, as fnordian had an additional change we're still discussing.
(related to issue #14059)
Reported by: fnordian
Patches:
chan_sip_hfield.patch uploaded by fnordian (license 110)
20090116__bug14059.diff.txt uploaded by Corydon76 (license 14)
Tested by: fnordian, Corydon76, oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #14185)
Reported by: Nick_Lewis
Patches:
chan_sip.c-expiryrequest6.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r172169 | oej | 2009-01-29 09:48:18 +0100 (Tor, 29 Jan 2009) | 16 lines
Make sure that we always add the hangupcause headers. In some cases, the owner was disconnected before we checked for the cause.
This patch implements a temporary storage in the pvt and use that instead.
The code is based on ideas from code from Adomjan in issue #13385 (Add support for Reason: header)
Thanks to Klaus Darillion for testing!
(closes issue #14294)
related to issue #13385
Reported by: klaus3000 and adomjan
Patches:
bug14294b.diff uploaded by oej (license 306)
Based on 20080829_chan_sip.c-q850reason_header.patch uploaded by adomjan (license 487)
Tested by: oej, klaus3000
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warning, breaking my dev-mode build. This fixes it.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171963 | tilghman | 2009-01-28 11:25:18 -0600 (Wed, 28 Jan 2009) | 2 lines
Clarify log message (suggested by manxpower on #asterisk-dev)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines
Fix devicestate problems for "always-on" agent channels
A revision to chan_agent attempted to "inherit" the device
state of the underlying channel in order to report the
device state of an agent channel more accurately.
The problem with the logic here is that it makes no sense to
use this for always-on agents. If the agent is logged in, then
to the underlying channel, the agent will always appear to be
"in use," no matter if the agent is on a call or not. The reason
is that to the underlying channel, the channel is currently in use
on a call to the AgentLogin application.
The most common cause that I found for this issue to occur was for
a SIP channel to be the underlying channel type for an Agent channel.
If the SIP phone re-registers, then the registration will cause the
device state core to query the device state of the SIP channel. Since the
SIP channel is in use, the Agent channel would also inherit this status.
Once the agent channel was set to "in use" there was no way that the device
state could change on that channel unless the agent logged out.
The solution for this problem is a bit different in 1.4 than it is in the
other branches. In 1.4, there will be a one-line fix to make sure that only
callback agents will inherit device state from their underlying channel type.
For the other branches of Asterisk, since callback support has been removed, there
is also no need for device state inheritance in chan_agent, so I will simply be
removing it from the code.
In addition, the 1.4 source is getting a new comment to help the next person who
edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be
used to determine if the agent is a callback agent or not.
(closes issue #14173)
Reported by: nathan
Patches:
14173.patch uploaded by putnopvut (license 60)
Tested by: nathan, aramirez
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(issue #14104)
Reported by: alecdavis
Tested by: dbailey
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Merged revisions 171527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171527 | oej | 2009-01-27 15:33:20 +0100 (Tis, 27 Jan 2009) | 13 lines
Use the same branch tag in CANCEL as in INVITE
Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now.
I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems.
Thanks Fredrik for pointing out where the bug in the SIP messaging was.
(closes issue #14346)
Reported by: oej
Patches:
bug14346.diff uploaded by oej (license 306)
Tested by: oej
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handle in the future as a group of settings for a group of devices.
At some point, I want one sip_cfg per domain handled, so we can have "group" settings.
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same place.
Default setting is set before we activate the channel or at reloads, not where we declare the variable.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171264 | oej | 2009-01-26 13:51:53 +0100 (MÃ¥n, 26 Jan 2009) | 9 lines
Don't retransmit 401 on REGISTER requests when alwaysauthreject=yes
(closes issue #14284)
Reported by: klaus3000
Patches:
patch_chan_sip_unreliable_1.4.23_14284.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines
Correctly track the hookstate
(closes issue #13686)
Reported by: itiliti
Patches:
20081013__bug13686.diff.txt uploaded by Corydon76 (license 14)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170588 | tilghman | 2009-01-23 13:20:44 -0600 (Fri, 23 Jan 2009) | 2 lines
Additions to AST-2009-001
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r170504 | file | 2009-01-23 14:04:08 -0400 (Fri, 23 Jan 2009) | 4 lines
Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold.
(closes issue #14295)
Reported by: klaus3000
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(closes issue #13182)
Reported by: pj
Patches:
2009011901_dontcleardisplay.diff.txt uploaded by mvanbaak (license 7)
Tested by: mvanbaak, pj
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prevented the call from being processed
(issue #14313)
Reported by: seandarcy
Tested by: dbailey
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PTMP connections
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Change configure script to detect the new ioctl call data structure.
(issue #14104)
Reported by: alecdavis
Patches:
mwiioctl_structure_asterisk.diff4.txt uploaded by dbailey (license )
Tested by: alecdavis, dbailey
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* Other minor changes.
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ccesario on IRC pointed out that his sip peers were not displayed
properly when he would issue the command "sip show peers." The problem
was that the onlymatchonip field was used to determine if the endpoint
was a "peer" or "user." The tricky part is that a "friend" is supposed
to be treated as both a "user" and a "peer" but the logic would not allow
"friends" to show up as "peers" since onlymatchonip was set to FALSE
for friends.
I have modified the sip_peer structure to more explicitly keep track of
what type endpoint it is so that the various manager and CLI commands
will display the expected information
Reported by ccesario via IRC
Tested by ccesario
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When an ast_str expands to hold more data, any pointers that were pointing
to the data prior to the expansion will be pointing at invalid memory. This
change makes such pointers used in chan_sip.c instead be offsets from the
beginning of the string so that the same math may be applied no matter where
in memory the string resides.
To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added
to chan_sip.c so that given a sip_request and an offset, the string at that
offset is returned.
(closes issue #14220)
Reported by: riksta
Tested by: putnopvut
Review http://reviewboard.digium.com/r/126/
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Dan saw regular segfaults with the old implementation and
rewrote it to make it really eventbased.
I altered it to be trunk compatible and wedhorn gave some feedback
and ideas how to make it even better.
(closes issue #13821)
Reported by: DEA
Patches:
chan_skinny-mwi-events.txt uploaded by DEA (license 3)
Tested by: mvanbaak, DEA
"no probs by me" from wedhorn
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determining the number of messages pending for MWI call
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ability.
(closes issue #14104)
Reported by: alecdavis
Patches:
asttrunk-14104.diff2.txt uploaded by dbailey (license )
chan_dahdi.rpas_and_fsk.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, dbailey
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r169210 | mmichelson | 2009-01-19 09:52:15 -0600 (Mon, 19 Jan 2009) | 13 lines
Prevent a crash in chan_local due to a potential NULL pointer dereference
Move the check for if both channels on a local_pvt have generators to below
where p->chan is checked for NULLity (NULLness?). This prevents a crash from
occurring if p->chan is NULL.
(closes issue #14189)
Reported by: sascha
Patches:
14189.patch uploaded by putnopvut (license 60)
Tested by: sascha
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This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up
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(closes issue #14233)
Reported by: chris-mac
Patches:
asterisk-bug14233.diff.txt uploaded by jamesgolovich (license 176)
Tested by: jamesgolovich, chris-mac, otherwiseguy
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r168975 | mmichelson | 2009-01-16 16:42:13 -0600 (Fri, 16 Jan 2009) | 18 lines
Account for possible NULL pointer when we receive a 408 in response to a REGISTER
It may be that by the time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the corresponding sip_pvt has
gone away. This situation was handled properly for a 200 OK response, but the 408
case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash
This commit fixes this assumption and prints out a message to the console if we should
receive a late 408 response to a REGISTER
(closes issue #14211)
Reported by: aborghi
Patches:
14211.diff uploaded by putnopvut (license 60)
Tested by: aborghi
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In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically
allocated buffer to hold the text of the request. There was a check in the
add_line function to not attempt to write the line into the buffer if we
did not have room for it.
In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str
structure is used to hold the text. Since it may grow to fit an arbitrarily
sized string, this check in add_line is no longer valid.
I found this oddity while attempting to fix issue #14220; however, I do not
believe that this is the fix for that issue since the output supplied by the
reporter did not contain the warning message that would be printed had this
condition been satisfied.
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source code warning.
Thanks Nick Lewis for pointing this out in the bug tracker.
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(closes issue #14246)
Reported by: klaus3000
Patches:
2patch_chan_sip_SIPRemoveHeader_trunk.txt uploaded by klaus3000 (license 65)
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