host portion in the Contact URI and specifies a
transport, the parsing done in parse_moved_contact
resulted in a malformed URI.
This commit fixes the parsing so that a proper
Dial string may be formed when the forwarded
call is placed.
(closes issue #13523)
Reported by: mattdarnell
Patches:
13523v2.patch uploaded by putnopvut (license 60)
Tested by: mattdarnell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #13524)
Reported by: wedhorn
Patches:
unload.diff uploaded by wedhorn (license 30)
With small tweak by me to prevent a crash
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
subscription if we have previously subscribed. Otherwise
a segfault will occur.
(closes issue #13476)
Reported by: jonnt
Patches:
13476.patch uploaded by putnopvut (license 60)
Tested by: jonnt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines
Create rules for disallowing contacts at certain addresses, which may
improve the security of various installations. As this does not change
any default behavior, it is not classified as a direct security fix for
anything within Asterisk, but may help PBX admins better secure their
SIP servers.
(closes issue #11776)
Reported by: ibc
Patches:
20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r142079 | mmichelson | 2008-09-09 11:19:17 -0500 (Tue, 09 Sep 2008) | 21 lines
When determining if codecs used by SIP peers allow
the media to be natively bridged, use the jointcapability
instead of the peercapability.
It seems that the intent of using the peercapability was to
expand the choice of codecs for the call to increase the
chances of being able to native bridge the channels. The
problem is that if a codec were settled on for the native
bridge and that wasn't a codec that was configured to be used
by Asterisk for that peer, then Asterisk would send a
REINVITE with no codecs in the SDP which is a bug no matter
how you slice it.
(closes issue #13076)
Reported by: ramonpeek
Patches:
13076.patch uploaded by putnopvut (license 60)
Tested by: tbelder
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r141809 | mmichelson | 2008-09-08 16:10:10 -0500 (Mon, 08 Sep 2008) | 14 lines
Fix pedantic mode of chan_sip to only check the
remote tag of an endpoint once a dialog has
been confirmed. Up until that point, it is possible
and legal for the far-end to send provisional
responses with a different To: tag each time. With
this patch applied, these provisional messages
will not cause a matching problem.
(closes issue #11536)
Reported by: ibc
Patches:
11536v2.patch uploaded by putnopvut (license 60)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line
This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Changes applied by this patch:
- Fix autocompletion in 'sip prune realtime', sip peers where never auto completed. Now we complete this command with:
'sip prune realtime peer' -> all | like | sip peers
Also I have modified the syntax in the usage, was wrong...
- Pass ast_cli_args->argv and ast_cli_args->argc while running autocompletion on CLI commands (CLI_GENERATE).
With this we avoid comparisons on ast_cli_args->line like this:
strcasestr(a->line, " description")
strcasestr(a->line, "descriptions ")
strcasestr(a->line, "realtime peer"), and so on..
Making the code more confusing (check the spaces in description!).
The only thing we must be sure is to first check a->pos or a->argc.
- Fix 'iax2 prune realtime' autocompletion, now we autocomplete this command with 'all' & 'iax2 peers', check a look that iax2 peers where all the peers, now only the ones in the cache..
(closes issue #13133)
Reported by: eliel
Patches:
clichanges.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This will print the subs and their status for every line (if any).
wedhorn did most of the work with his patch which introduced
'skinny show debug' but a discussion on IRC stated that it should be
added to 'skinny show lines'
Input on the output format by Qwell on IRC.
(closes issue #13344)
Reported by: wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug 2008) | 22 lines
After working on the ao2_containers branch, I noticed
something a bit strange. In all cases where we provide
a callback function to ao2_container_alloc, the callback
function would only return 0 or CMP_MATCH. After inspecting
the ao2_callback() code carefully, I found that if you're
only looking for one specific item, then you should return
CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue
traversing the current bucket until the end searching for
more matches.
In cases like chan_iax2 where in 1.4, all the peers are
shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be
traversed even if the peer is one of the first ones come
across in the bucket.
All the changes I have made were for cases where the
callback function defined was passed to ao2_container_alloc
so that calls to ao2_find could find a unique instance
of whatever object was being stored in the container.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r140417 | mmichelson | 2008-08-29 10:26:52 -0500 (Fri, 29 Aug 2008) | 10 lines
Fix SIP's parsing so that if a port is specified
in a string to Dial(), it is not ignored.
(closes issue #13355)
Reported by: acunningham
Patches:
13355v2.patch uploaded by putnopvut (license 60)
Tested by: acunningham
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140418 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug 2008) | 11 lines
Fix tag checking in get_sip_pvt_byid_locked when
in pedantic mode. The problem was that the wrong
tags would be compared depending on the direction
of the call.
(closes issue #13353)
Reported by: flefoll
Patches:
chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r140060 | russell | 2008-08-26 11:07:58 -0500 (Tue, 26 Aug 2008) | 6 lines
Fix some bogus scheduler usage in chan_sip. This code used the return value
of a completely unrelated function to determine whether the scheduler should
be run or not. This would have caused the scheduler to not run in cases where
it should have. Also, leave a note about another scheduler issue that needs
to be addressed at some point.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r140051 | russell | 2008-08-26 10:27:23 -0500 (Tue, 26 Aug 2008) | 15 lines
Fix a race condition with the IAX scheduler thread. A lock and condition are
used here to allow newly scheduled tasks to wake up the scheduler just in case
the new task needs to run sooner than the current wakeup time when the thread
is sleeping. However, there was a race condition such that a newly scheduled
task would not properly wake up the scheduler or affect the wake up period.
The order of execution would have been:
1) Scheduler thread determines wake up time of N ms.
2) Another thread schedules a task and signals the condition, with an
execution time of < N ms.
3) Scheduler thread locks and goes to sleep for N ms.
By moving the sleep time determination to inside the critical section, this
possibility is avoided.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
headers for the SipNotify manager command was
causing the current manager session to become
disconnected. Change the return value to 0 for
these cases.
Also change a test for a NULL pointer to be
ast_strlen_zero instead.
(closes issue #13351)
Reported by: Laureano
Patches:
sipnotify_action_fix.patch uploaded by Laureano (license 265)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r138942 | mmichelson | 2008-08-19 18:17:17 -0500 (Tue, 19 Aug 2008) | 11 lines
Reset agent_pvt variables back to the values in agents.conf
(from what the corresponding channel variables were set to)
when the agent logs out.
(closes issue #13098)
Reported by: davidw
Patches:
20080731__issue13098_agent_ackcall_not_reset.diff uploaded by bbryant (license 36)
Tested by: davidw
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/tex/misdn.tex
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Christian Richter
Mark Michelson
Michiel van Baak
Steve Murphy
Tilghman Lesher
Jeff Peeler
Sean Bright
Doug Bailey
Russell Bryant
Terry Wilson
Philippe Sultan
Jason Parker
Richard Mudgett