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${ noResults }
24833 Commits (ad0e70ba83d3b7d08127b0a32b73d977dff0dac5)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Joshua Colp
|
d183c6e134 |
Return a channel snapshot when originating using ARI, and subscribe the Stasis application to it.
This change allows a user of ARI to know what channel it has originated and also follow any progress. If a Stasis application is provided it will be automatically subscribed to the originated channel immediately. (closes issue ASTERISK-22485) Reported by: David Lee Review: https://reviewboard.asterisk.org/r/2910/ ........ Merged revisions 401281 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401282 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Richard Mudgett
|
a80a6a7631 |
res_parking: Remove setting useless flag.
........ Merged revisions 401271 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401272 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
David M. Lee
|
14dc5776d2 |
This is just a quick script for dumping swagger-ui into static-http,
so that it can be served by the Asterisk web server. I had to change the Makefile in order to recursively install content from the static-http directory, hence the code review instead of just putting it in. Review: https://reviewboard.asterisk.org/r/2924/ ........ Merged revisions 401261 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401263 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Mark Michelson
|
c30170d9a2 |
Resolve some memory leaks due to incorrect for loop / ao2 ref usage.
A common idiom in Asterisk is to due something like:
for (ao2_obj = list_beginning; ao2_obj = next_item; ao2_ref(ao2_obj, -1)) {
...do stuff...
}
This is nice because it automatically takes care of the object references
for you. However, there is a pitfall here. If a break statement is in the
for loop, then the current reference is not cleaned up. In some cases, this
is on purpose, but in others there is a leak. This commit fixes the leak
cases.
........
Merged revisions 401248 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
12 years ago |
|
Richard Mudgett
|
057d105c5a |
Add channel lock protection around translation path setup.
Most callers of ast_channel_make_compatible() happen before the channels enter a two party bridge. With the new bridging framework, two party bridging technologies may also call ast_channel_make_compatible() when there is more than one thread involved with the two channels. * Added channel lock protection in set_format() and ast_channel_make_compatible_helper() when dealing with the channel's native formats while setting up a translation path. * Fixed best_src_fmt and best_dst_fmt usage consistency in ast_channel_make_compatible_helper(). The call to ast_translator_best_choice() got them backwards. * Updated some callers of ast_channel_make_compatible() and the function documentation. There is actually a difference between the two channels passed in. * Fixed the deadlock potential in res_fax.c dealing with ast_channel_make_compatible(). The deadlock potential was already there anyway because res_fax called ast_channel_make_compatible() with chan locked. (closes issue ASTERISK-22542) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2915/ ........ Merged revisions 401239 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401240 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Richard Mudgett
|
7352433be5 |
Tweak ast_bridge_depart() doxygen.
........ Merged revisions 401232 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401233 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Mark Michelson
|
db099a641b |
Remove the bit about requiring ast_bridge_depart() to be called before ast_bridge_destroy().
........ Merged revisions 401223 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401224 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Mark Michelson
|
5bba145ee1 |
Clarify in ast_bridge_destroy() about how departable channels must be handled.
........ Merged revisions 401212 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Michael L. Young
|
32d758ed32 |
Remove Port Restriction When Checking For NAT
When trying to determine if a peer is behind NAT, we should not be using the
ports when comparing addresses.
This patch removes the port from being checked and just useds the addresses
now.
(closes issue ASTERISK-22729)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-remove-using-port-for-nat-check.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2927/
........
Merged revisions 401182 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401183 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
Walter Doekes
|
f33e0776ec |
Properly copy/remove the device state cache flag over a masquerade.
In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells the devstate system to not cache states for non-real devices. However, when optimizing away channels (ast_do_masquerade), that flag wasn't copied. In my case, using Local devices as queue members created a situation where the endpoint was considered in use, but the state change of the device being available again was ignored (not cached). The endpoint channel was optimized into the (previously) Local channel, but kept the do-not-cache flag. The end result being that the queue member apparently stayed in use forever. (closes issue ASTERISK-22718) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/2925/ ........ Merged revisions 401178 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401179 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401180 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401181 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Michael L. Young
|
42f3cae1fd |
Fix Setting A chan_sip Dialog's SIP_NAT_FORCE_RPORT Flag
A condition was added in a commit to fix ASTERISK-21374, that, if the
SIP_PAGE3_NAT_AUTO_RPORT flag was set, to then copy a peer's SIP_NAT_FORCE_RPORT
flag to the dialog. This condition should not have been there since it assumed
that if Asterisk is in an environment where NAT is involved, that the auto_* nat
settings or force_rport setting would be on in the global settings. If the nat
setting in the global setting is set to 'nat=no' and then turned on for peers
(which is not quite the recommended way, although it is allowed) this flag is
never copied to the dialog resulting in problems like, REGISTER replies going
to the wrong port.
This patch removes this conditional check and will now always use the peer's
flag which by this point in the code the checks on whether the peer is behind
NAT or not (if using auto_force_rport) have already been run.
(closes issue ASTERISK-22236)
Reported by: Filip Frank
Tested by: Michael L. Young
Patches:
asterisk-2236-always-set-rport.diff uploaded
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2919/
........
Merged revisions 401167 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 401168 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
Jonathan Rose
|
9a45173715 |
res_parking: Fix bug where reloading immediately wipes new parkpos extensions
(closes issue ASTERISK-22631) Reported by: Kevin Harwell ........ Merged revisions 401158 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Kinsey Moore
|
4cdd02ce26 |
Reduce log level of a non-pubsub error message
Drop an error log message to debug level 1 since distributed device
state functions correctly when receiving this message and it spams the
logs.
(closes issue ASTERISK-22410)
Reported by: abelbeck
Patches:
asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
........
Merged revisions 401119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 401120 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 401121 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
|
Richard Mudgett
|
868a76f897 |
ARI: Fix crash when POST /playback/{id}/control does not have an operation parameter.
(closes issue ASTERISK-22680) Reported by: John Bigelow ........ Merged revisions 401107 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
David M. Lee
|
3ce661a751 |
Oops. Leftover /stasis reference
........ Merged revisions 401096 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401097 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Kinsey Moore
|
2fc0a8873c |
Clarify documentation for channel and bridge list
This makes it clear that the ARI API calls for listing channels and bridges will list all channels or bridges in the system and not just those that are in or are controlled by a Stasis application. (closes issue ASTERISK-22635) Reported by: Kevin Harwell ........ Merged revisions 401087 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Walter Doekes
|
e694a976f6 |
Don't check all realtime queues when doing "queue show some_queue".
When using realtime queues, queues have to be fetched from the database every now and then to see if any info has been changed or to see if the queue has been removed. When fetching info for an individual queue, the pruning of other queues is unnecessarily costly. Review: https://reviewboard.asterisk.org/r/2907/ ........ Merged revisions 401049 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401076 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401077 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401079 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Paul Belanger
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7955e89d06 |
Use POST / DELETE to toggle ARI bridge moh
Review: https://reviewboard.asterisk.org/r/2911/ ........ Merged revisions 401040 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Richard Mudgett
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ef7c5a04c0 |
translate.c: Some minor code tweaks.
* Consistently compare format2index() return value so matrix_get() cannot get passed negative values. * Optimize ast_translator_best_choice() to defer initializing things until needed. Also cached the matrix_get() return value rather than repeatedly calling it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Richard Mudgett
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66e7dc041d |
bridge_native_dahdi: Return channel join failure if could not make the channels compatible.
........ Merged revisions 401030 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401031 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Richard Mudgett
|
3208058831 |
chan_iax2: Fix channel left locked in off nominal code path.
........ Merged revisions 401016 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401017 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401020 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Kinsey Moore
|
ec83706003 |
Ensure bridge record error responses validate
This adds the list of expected errors to the /bridges/{bridgeId}/record
ARI documentation so that outbound 4xx errors validate properly.
Previously, this would result in a response validation failure.
(closes issue ASTERISK-22627)
Reported by: Joshua Colp
........
Merged revisions 401018 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
|
Paul Belanger
|
6072e043cf |
Use POST / DELETE to toggle hold / moh for ARI channels
This change updates how we handle toggle events, rather then create two different function names, we'll just use POST / DELETE from HTTP to handle it. Review: https://reviewboard.asterisk.org/r/2906/ ........ Merged revisions 400999 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Mark Michelson
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2c927b871f |
Prevent chan_sip from sending duplicate BYEs.
When a 200 OK for an initial INVITE is received, we were doing the right thing by ACKing and sending an immediate BYE. However, we also were doing the wrong thing and queuing an answer frame, thus causing the call to be answered. This would cause the call to be hung up by the channel thread, thus resulting in a second BYE being sent out. In this fix, I also have set the hangupcause to be correct since the initial BYE being sent by Asterisk had an unknown hangup cause. I have changed to using "Bearer capabilty not available" since the call was hung up due to an SDP offer/answer error. (closes issue ASTERISK-22621) reported by Kinsey Moore ........ Merged revisions 400970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400971 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400984 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400998 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
David M. Lee
|
55e15a2cc9 |
My doc correction in r400842 had a silly bug.
Because I added a wiki_description to models and not their properties, the rendered wiki page had the model description instead of the property descriptions, which looks very silly indeed. (closes issue ASTERISK-22705) ........ Merged revisions 400958 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Richard Mudgett
|
2127848d6c |
chan_dahdi: Add config support for hwgain settings.
* Add hwtxgain and hwrxgain config options to chan_dahdi.conf with
documentation in chan_dahdi.conf.sample.
(closes issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch uploaded by rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
|
Richard Mudgett
|
edb437ee6c |
chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" output.
* Remember the swgain setting from CLI "dahdi set swgain" command so the
CLI "dahdi show channel" output will reflect the current setting.
* Updated CLI "dahdi set hwgain" and "dahdi set swgain" documentation.
(issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded by rmudgett
........
Merged revisions 400907 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 400909 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 400911 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
|
Mark Michelson
|
47e910bfe6 |
chan_sip: Do not increment the SDP version between 183 and 200 responses.
Bumping the SDP version number can cause interoperability problems since receivers of the responses will expect that a 200 SDP will be identical to a previous 183 SDP. (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches: dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418) ........ Merged revisions 400906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400908 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400910 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Kevin Harwell
|
61b60fab2d |
pjsip outbound registration: Log message says received a 408 when we didn't
If the server didn't exist that we are trying to register to the log message would say that a 408 was received from that server when in reality one wasn't. Added log messages stating no response was received if the response does not exist. (closes issue ASTERISK-22554) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2893/ ........ Merged revisions 400890 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400891 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Matthew Jordan
|
fd4919e466 |
Remove duplicate module info block
The module info block was repeated twice. Once is sufficient. ........ Merged revisions 400881 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Joshua Colp
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b47851264e |
Fix a race condition in res_pjsip_session with rapidly terminating the session.
The INVITE session state callback wrongly assumes that a session will always exist, but when rapidly terminating the session this assumption goes out the window. As all handler code for the INVITE session state callback requires the session it will now just exit immediately if no session exists. (closes issue ASTERISK-22668) Reported by: John Bigelow ........ Merged revisions 400872 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Kinsey Moore
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1a0a2b3e4c |
Fix realm comparison for outbound auth
When generating the list of authentication credentials to pass to PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not always NULL-terminated. This sometimes resulted in incorrect text for the realm and a failure to match the realm for authentication purposes which was causing the outbound nominal auth pjsip basic call test to bounce. This now uses the pj_str_t that contains the realm instead of generating a new one. Thanks to John Bigelow for helping to narrow this down. ........ Merged revisions 400863 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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224f44c971 |
channel.h: whitespace changes.
........ Merged revisions 400854 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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David M. Lee
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9234804a3b |
Multiple revisions 400508,400842-400843,400848
........
r400508 | dlee | 2013-10-03 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line
Corrected response class for stopPlayback
........
r400842 | dlee | 2013-10-10 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line
Correct some ARI wiki rendering errors
........
r400843 | dlee | 2013-10-10 14:26:19 -0500 (Thu, 10 Oct 2013) | 1 line
Updated /play resource docs. The playback of http: resources isn't implemented... yet
........
r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5 lines
Fix a stupid copy/paste error in ARI docs.
Patches:
ari-doc-patch.txt uploaded by jbigelow (license 5091)
........
Merged revisions 400508,400842-400843,400848 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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David M. Lee
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07e2b1d028 |
Fixed merge tracking for r400360, which was somehow lost
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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0ddcee5a46 |
Softmix: Fix crash when switching from softmix to another bridge technology.
The crash is caused by a race condition when switching between native RTP
and softmix bridging technologies. In this situation, the bridging
technology is switched from native RTP to softmix, and then back to native
RTP fast enough that the softmix private data gets destroyed before the
softmix mixing thread gets started.
Thanks to Kinsey Moore for the crash analysis.
* Fix race condition when starting the softmix mixing thread and switching
to another bridge technology.
(closes issue ASTERISK-22678)
Reported by: John Bigelow
Patches:
jira_asterisk_22678_v12.patch (license #5621) patch uploaded by rmudgett
Tested by: John Bigelow
........
Merged revisions 400849 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
|
Joshua Colp
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113c281782 |
Perform validation of permanent contacts on AORs in res_pjsip.
........ Merged revisions 400833 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Joshua Colp
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cbbcf1808c |
Fix an assertion in res_pjsip when specifying an invalid outbound proxy.
This change fixes two issues when setting an outbound proxy: 1. The outbound proxy URI was not parsed and validated during configuration. 2. If an outgoing dialog was created and the outbound proxy could not be set an assertion would occur because the usage count on the dialog was not decremented. The documentation has also been updated to specify that a full URI must be specified for the outbound proxy. (closes issue ASTERISK-22672) Reported by: Antti Yrjola ........ Merged revisions 400824 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400825 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Matthew Jordan
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fcf6c84666 |
Use 'z' as the format specifier for size_t
Using 'lu' will produce a compiler warning for some versions of gcc and on some architectures. 'z' should be portable as a format specifier for size_t. ........ Merged revisions 400812 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400813 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Matthew Jordan
|
b36ef0b412 |
Add PJSIP_HEADER function for manipulation of SIP headers in the PJSIP stack
This patch adds support to the PJSIP stack in Asterisk for SIP header manipulation. Note that this is analagous to SIPAddHeader/SIPRemoveHeader. For PJSIP_HEADER, an incoming supplemental session callback is registered that takes the pjsip_hdrs from the incoming session and stores them in a linked list in the session datastore. Calls to PJSIP_HEADER traverse over the list and return the nth matching header where 'n' is the 'number' argument to the function. When adding a header, the first call creates a datastore and linked list and adds the datastore to the session. The header is then created as a pjsip_hdr and added to the list. An outgoing supplemental session callback then traverses the list and adds the headers to the outgoing pjsip_msg. When removing a header, the list created with PJSIP_HEADER(add,...) is traversed and all matching entries are removed. (closes issue ASTERISK-22498) Reported by: George Joseph patch: res_pjsip_header_funcs_v1.patch uploaded by george.joseph (License 6322) ........ Merged revisions 400771 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Kinsey Moore
|
8021958099 |
Add warning when compiling with iODBC support
When running configure, libiodbc2 development headers will fulfill the
requirement for ODBC development headers, but will not function
properly. This adds a warning when libiodbc2 development headers are
detected instead of unixodbc development headers.
(closes issue ASTERISK-22459)
Reported by: Patrick Maille
Tested by: Walter Doekes
Patches:
issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes (License 5674)
........
Merged revisions 400767 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 400768 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400769 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Richard Mudgett
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478c88991e |
app_agent_pool: Fix AMI/CLI AgentLogoff soft preventing agents from logging back in.
* Clear the deferred_logoff flag when an agent logs in. (closes issue ASTERISK-22669) Reported by: John Bigelow ........ Merged revisions 400754 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Mark Michelson
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2904a198d5 |
Switch from using pjsip_strerror to pj_strerror.
pjsip_strerror is only aware of PJSIP-specific error codes. pj_strerror() is aware of all PJProject error codes and OS-specific error codes. This specifically fixes an oft-seen error in transport configuration code where EADDRINUSE would result in "Unknown PJSIP error 120098" instead of a useful message. ........ Merged revisions 400749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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f87086b374 |
app_confbridge: Can now set the language used for announcements to the conference.
ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in
confbridge.conf or by the dialplan function
CONFBRIDGE(bridge,language)=en.
(closes issue ASTERISK-19983)
Reported by: Jonathan White
Patches:
M19983_rev2.diff (license #5138) patch uploaded by junky (modified)
Tested by: rmudgett
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Merged revisions 400741 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400742 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Richard Mudgett
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665ef4c654 |
app_confbridge: Fix duplicate default_user profile.
* Fixed looking in the wrong profiles container to see if the default_user profile is already created in verify_default_profiles(). The bridge profile container is never going to hold user profiles. :) ........ Merged revisions 400723 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400724 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400728 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Kinsey Moore
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4e72ad1b4f |
Fix func_config list entry allocation
The AST_CONFIG dialplan function defined in func_config.c allocates its config file list entries using ast_malloc. List entry allocations destined for use with Asterisk's linked list API must be ast_calloc()d or otherwise initialized so that list pointers are set to NULL. These uses of ast_malloc have been replaced by ast_calloc to prevent dereferencing of uninitialized pointer values when traversing the list. (closes issue ASTERISK-22483) Reported by: Brian Scott ........ Merged revisions 400694 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400697 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400701 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Kinsey Moore
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4873c11f64 |
Fix STUN crash when using IPv6 any address
Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4
candidates are also added.
(closes issue ASTERISK-21917)
Reported by: Torrey Searle
Patches:
0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334)
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Merged revisions 400681 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400682 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Mark Michelson
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a4d6f43962 |
Push CLI qualify into the threadpool.
If you run Asterisk in the background and then connect to it through a separate console, the thread that runs CLI commands is not registered with PJLIB. Thus PJLIB does not like it when you attempt to send OPTIONS requests from that thread. So now we push the task into the threadpool, which we know to be registered with PJLIB. Thanks to Antti Yrjola for reporting this. ........ Merged revisions 400680 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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8eec8fbf83 |
Make app_queue and res_agi independent of AMI being enabled.
The https://reviewboard.asterisk.org/r/2888/ review changes manager to not subscribe to stasis when it is disabled for performance reasons. When manager is disabled app_queue and res_agi decline to load and fail to clean up what they have already allocated. * Made app_queue and res_agi clean up allocated resources when they decline to load. * Made app_queue and res_agi use their own subscriptions to the stasis topics instead of borrowing manager's message router structure inappropriately. (closes issue ASTERISK-22604) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/2902/ ........ Merged revisions 400671 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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96d27333d2 |
Miscellaneous stand alone comment cleanups.
........ Merged revisions 400661 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400662 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |