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${ noResults }
20423 Commits (0a02e9bcc1600a8c4ef67ab3bb47523548696d8d)
Author | SHA1 | Message | Date |
---|---|---|---|
|
3442f13da4 |
Merged revisions 277625 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277625 | tringenbach | 2010-07-16 17:43:39 -0500 (Fri, 16 Jul 2010) | 9 lines Save and restore AST_FLAG_BRIDGE_HANGUP_DONT on attended transfer. ast_bridge_call() clears AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer, ast_bridge_call() is called for a second bridge on the same channel, and it clears that flag, which still needs to get set for when the original ast_bridge_call() gets control back and checks it. Review: https://reviewboard.asterisk.org/r/741 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
5150954d4a |
Merged revisions 277497 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277497 | mnicholson | 2010-07-16 16:18:38 -0500 (Fri, 16 Jul 2010) | 4 lines Default to no udptl error correction so that error correction will be disabled in the event that the remote end indicates that they do not support the error correction mode we requested. FAX-128 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
5b8a8fc6c8 |
Fix reporting estimated queue hold time.
Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes. (closes issue #17498) Reported by: corruptor Patches: holdesecs_bug.diff uploaded by corruptor (license 253) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277488 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
fe9e0e672e |
Finally, a method that really fixes the assertions in chan_iax2.c related to cancelling lagid.
No, replacing usleep(1) with sched_yield() did not have an effect. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277484 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
34bc4b1dcb |
Merged revisions 277419 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277419 | rmudgett | 2010-07-16 15:18:54 -0500 (Fri, 16 Jul 2010) | 15 lines priexclusive in chan_dahdi.conf ignored when reloading dahdi module During a reload, the priexclusive and outsignalling parameters are not read in from the config file as intended. Unfortunately, they get set to defaults as a result. This patch makes sure that they do not get set to defaults during a reload. (closes issue #17441) Reported by: mtryfoss Patches: issue17441_v1.4.patch uploaded by rmudgett (license 664) issue17441_v1.6.2.patch uploaded by rmudgett (license 664) issue17441_trunk.patch uploaded by rmudgett (license 664) Tested by: rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f12afaaebc |
Add documentation for MOH realtime fields
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1fd618de5b |
updated devicestate test for device state changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277409 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b73c1377e5 |
Add missing handling for ringing state for use with queue empty options.
(closes issue #17471) Reported by: jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1c848835aa |
Merged revisions 277327 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277327 | mnicholson | 2010-07-16 13:30:22 -0500 (Fri, 16 Jul 2010) | 8 lines Interpret device state AST_DEVICE_UNKNOWN as extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035) Reported by: francesco_r Patches: pbx.c.patch uploaded by viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d72336e83f |
Merged revisions 277261 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines If variable gotten is not set, will segfault on Solaris. (closes issue #17636) Reported by: bklang ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277263 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e16a5e4727 |
Print f->subclass.integer instead of f->subclass.
(fix build breakage introduced in r277250) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d787ccff35 |
Merged revisions 277247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277247 | mnicholson | 2010-07-16 12:29:57 -0500 (Fri, 16 Jul 2010) | 4 lines For pass through DTMF tones, measure the actual duration between the begin and end packets on the wire. If it is detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf emulation. AST-362 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277250 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8eb9e0b938 |
Merged revisions 277182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines Total analysis time error with SIP and silence suppression When using app_amd with SIP providers that have silence suppression on, the iTotalTime count increases exponentially. (closes issue #17656) Reported by: juls ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2289649901 |
Fix up some weird indentation problems in reqresp_parser.c
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277175 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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215fb1ab9f |
Avoid crashing when installing a duplicate translation path with a lower cost.
(closes issue #17092) Reported by: moy Patches: translate.rev254273.patch uploaded by moy (license 222) Tested by: moy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277143 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0bb1fbb540 |
Add Despegar.com (my main sponsor) to the CREDITS file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277103 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5a1ed1f070 |
Formatting changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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93373d7bdf |
Formatting fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277065 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a3dd1d2188 |
Clarify syntax changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fcae566e94 |
Adding a few more to the list of CREDITS
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277027 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cbe0a6dc02 |
Formatting changes (guideline corrections)
Found a unused bag of curly brackets under my table. I always wondered where they had gone. They where indeed needed in chan_sip.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276989 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8c0ab98786 |
Adding a few more credits
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e129b31fc6 |
Add ability to configure the Max-Forwards header in the dialplan, as well as in
sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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65203b12dd |
Add a dialplan function to check if a queue exists: QUEUE_EXISTS
Review: https://reviewboard.asterisk.org/r/777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276950 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7515e03c8b |
And yet one more
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276911 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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be7fbdf25d |
"Item may be used uninitialized in this function."
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276910 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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dfba265a0b |
Fix reversed logic of if statement.
Found based on message from Philip Prindeville on the Asterisk Developers mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276909 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a76c8e36f5 |
Detect the --dynamic-list flag a bit better
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0ab4420d66 |
Fix build on FreeBSD
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276871 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e990f89a0e |
Fix trunk build for Mac OS X 10.6
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276870 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3cd3824171 |
Allow ipaddress to contain the maximum IPv6 address.
Also, update meetme to the full list of supported fields. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3299f13d94 |
Quote AC_SUBST within m4_ifval, so it does not get prematurely expanded.
(closes issue #17654) Reported by: pprindeville Patches: issue17654.diff uploaded by qwell (license 4) Tested by: qwell, pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276830 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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44ae0033be |
Correct not setting the bindport before attempting to open the socket.
Related to changes from 276571, I was accidentally testing with a port set in my configuration causing me to miss this. Also moved the TCP handling as well to occur before build_peer is called. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276788 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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753fa3b278 |
Define LLONG_MAX on systems that do not have it.
(closes issue #17644) Reported by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276769 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e2ff55122d |
Fix linking asterisk on CentOS 5, which is using gcc 4.1.1. Gcc 4.1.2 has the real fix.
Review: https://reviewboard.asterisk.org/r/790/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e7591ab428 |
Merged revisions 276652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r276652 | jpeeler | 2010-07-15 08:48:58 -0500 (Thu, 15 Jul 2010) | 2 lines In a perfect world, the frame source would never be NULL. In the meantime, don't crash when it is. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276653 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9778856a03 |
Add lua5.1 to the handy dandy list of packages.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276616 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2b2a6123de |
Fix MWI notification transmission problems over SIP.
MWI updates were not being sent if no messages were found in the event cache. This was corrected since a phone may need to clear its MWI status configured previously from another mailbox. Upon module or sip reload, MWI updates could not be sent due to the sipsock socket not being set early enough in reload_config. The code handling the descriptor assignment and such has simply been moved before the call to build_peer. Issuing a sip reload cleared the IP address of the peer, but skipped checking the database for registration information. The database is now checked both for sip reload and actually reloading the module. If a transmission occurs before the do_monitor thread has started, do not attempt to send a signal to it. (closes issue #17398) Reported by: ip-rob git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276571 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1e8c66e749 |
Fix errors where incorrect address information was printed.
ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions) uses thread-local storage for storing the string that it creates. In cases where ast_sockaddr_stringify_fmt was being called twice within the same statement, the result of one call would be overwritten by the result of the other call. This usually was happening in printf-like statements and was resulting in the same stringified addressed being printed twice instead of two separate addresses. I have fixed this by using ast_strdupa on the result of stringify functions if they are used twice within the same statement. As far as I could tell, there were no instances where a pointer to the result of such a call were saved anywhere, so this is the only situation I could see where this error could occur. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276570 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5d9aa45721 |
Make compile again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276531 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4c94d1ee23 |
Oops, merge reverted this fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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832d1296c6 |
Remove the old stub files, preferring the optional_api method.
(closes issue #17475) Reported by: tilghman Review: https://reviewboard.asterisk.org/r/695/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8e7d01d484 |
Don't try to call an embedded module's backup_globals() function until
after confirming it exists. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276441 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d1c9a4b366 |
handle special case were "200 Ok" to pending INVITE never receives ACK
Unlike most responses, the 200 Ok to a pending INVITE Request is acknowledged by an ACK Request. If the ACK Request for this Response is not received the previous behavior was to immediately destroy the dialog and hangup the channel. Now in an effort to be more RFC compliant, instead of immediately destroying the dialog during this special case, termination is done with a BYE Request as the dialog is technically confirmed when the 200 Ok is sent even if the ACK is never received. The behavior of immediately hanging up the channel remains. This only affects how dialog termination proceeds for this one special case. RFC 3261 section 13.3.1.4 "If the server retransmits the 2xx response for 64*T1 seconds without receiving an ACK, the dialog is confirmed, but the session SHOULD be terminated. This is accomplished with a BYE, as described in Section 15." git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276439 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cf7bbcc4c6 |
Expand the caller ANI field to an ast_party_id
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e2599bc42c |
collapse debug code in retrans_pkt into separate lines
I've been working in this function a bunch lately, and these huge debug strings are getting annoying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276392 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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2cf60bb09d |
Make compile again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276391 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f4c665ee13 |
Do not skip sending MWI for a peer if an address is defined. Really just a merge mistake from IPv6
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276389 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e19a6c248f |
Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel.
Change the documented pgsql schema to use "timestamp" instead of "time", as the latter is only a time without a date. Added some missing columns for cel's pgsql schema, and corrected spelling on some others. Updated cel's uniqueid size to be the same as the cdr. Added id column to cel's pgsql schema and updated code to allow unknown columns to get their default value instead of forcing 0 or empty string. Added microseconds to the timestamp cel logs to pgsql. Review: https://reviewboard.asterisk.org/r/734 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ec37ffbdaf |
ast_callerid restructuring
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |