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populate the FAXOPT output variables.
FAX-222
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280058 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26 Jul 2010) | 13 lines
Apply all patches in:
https://issues.asterisk.org/view.php?id=13573
(closes issue #13573)
Reported by: navkumar
Patches:
res_config_ldap-category.diff uploaded by navkumar (license 580)
res_config_ldap.patch uploaded by bencer (license 961)
res_config_ldap uploaded by bencer (license 961)
Tested by: suretec
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279113 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278957 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/798/
Thanks Mark for a quick review!
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Review: https://reviewboard.asterisk.org/r/793/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278538 f38db490-d61c-443f-a65b-d21fe96a405b
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This code did not properly check FD_SETSIZE to ensure that it did not try to
select() on fds that were too large. Switching to poll() removes the limitation
on the maximum fd value.
(closes issue #15915)
Reported by: keiron
(closes issue #17187)
Reported by: Eddie Edwards
(closes issue #16494)
Reported by: Hubguru
(closes issue #15731)
Reported by: flop
(closes issue #12917)
Reported by: falves11
(closes issue #14920)
Reported by: vrban
(closes issue #17199)
Reported by: aleksey2000
(closes issue #15406)
Reported by: kowalma
(closes issue #17438)
Reported by: dcabot
(closes issue #17325)
Reported by: glwgoes
(closes issue #17118)
Reported by: erikje
possibly other issues, too ...
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actual failure.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278426 f38db490-d61c-443f-a65b-d21fe96a405b
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to be used, even when realtime is used.
(closes issue #17082)
Reported by: coolmig
Patches:
20100720__issue17082.diff.txt uploaded by tilghman (license 14)
Tested by: coolmig
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278307 f38db490-d61c-443f-a65b-d21fe96a405b
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handled.
FAX output channel variables will now match the values reported by FAXOPT() and should be set in all failure and success cases.
This commit also contains a few modifications to the way FAXOPT() variables are populated in a few spots and fixes for some reference count leaks of the session details structure in some failure cases.
Also found and fixed more cases where FAXOPT(status) may not have gotten set.
FAX-214
FAX-203
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r277568 | tilghman | 2010-07-16 16:54:29 -0500 (Fri, 16 Jul 2010) | 8 lines
Since we split values at the semicolon, we should store values with a semicolon as an encoded value.
(closes issue #17369)
Reported by: gkservice
Patches:
20100625__issue17369.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277773 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issues #17667)
Reported by: snuffy
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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: http://svn.digium.com/svn/asterisk/trunk@276570 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17475)
Reported by: tilghman
Review: https://reviewboard.asterisk.org/r/695/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276490 f38db490-d61c-443f-a65b-d21fe96a405b
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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/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276391 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
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tracking down the source.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275105 f38db490-d61c-443f-a65b-d21fe96a405b
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This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
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midterm evaluation.
Review: https://reviewboard.asterisk.org/r/757/
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possible. Previously some failure cases did not result in proper FAXOPT values.
FAX-203
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dialplan function. Also make fax_rate_str_to_int() return an unsigned int and return 0 instead of -1 in the event of an error.
FAX-202
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274243 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r274157 | mmichelson | 2010-07-06 09:29:23 -0500 (Tue, 06 Jul 2010) | 16 lines
Fix problem with RFC 2833 DTMF not being accepted.
A recent check was added to ensure that we did not erroneously
detect duplicate DTMF when we received packets out of order.
The problem was that the check did not account for the fact that
the seqno of an RTP stream will roll over back to 0 after hitting
65535. Now, we have a secondary check that will ensure that the
seqno rolling over will not cause us to stop accepting DTMF.
(closes issue #17571)
Reported by: mdeneen
Patches:
rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
Tested by: richardf, maxochoa, JJCinAZ
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274164 f38db490-d61c-443f-a65b-d21fe96a405b
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(Also fix the typos in the comments)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273641 f38db490-d61c-443f-a65b-d21fe96a405b
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FAX-177
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273464 f38db490-d61c-443f-a65b-d21fe96a405b
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Also clean up some coding errors.
(closes issue #17469)
Reported by: wdoekes
Patches:
astsvn-rtp-set-debug-ip.patch uploaded by wdoekes (license 717)
Tested by: wdoekes, pabelanger
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273233 f38db490-d61c-443f-a65b-d21fe96a405b
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the config file also changes.
(closes issue #16982)
Reported by: dmitri
Patches:
res_musiconhold.patch uploaded by dmitri (license 1001)
Tested by: atis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272332 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17215)
Reported by: vazir
Patches:
20100518__issue17215.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272260 f38db490-d61c-443f-a65b-d21fe96a405b
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Don't Finalize() if Initialize() did not succeed. This resulted in an error
about trying to Finalize() an invalid handle.
Also trim some trailing whitespace while in the area.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271867 f38db490-d61c-443f-a65b-d21fe96a405b
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Using this method makes it so res_fax doesn't have to be rebuilt on every
svn update.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines
Do not use sizeof to calculate size of a heap allocated character array.
Change left out from 271399.
(closes issue #16053)
Reported by: diLLec
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271551 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines
Fix crash when parsing some heavily nested statements in AEL on reload.
Due to the recursion used when compiling AEL in gen_prios, all the stack space
was being consumed when parsing some AEL that contained nesting 13 levels deep.
Changing a few large buffers to be heap allocated fixed the crash, although I
did not test how many more levels can now be safely used.
(closes issue #16053)
Reported by: diLLec
Tested by: jpeeler
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(closes issue #16153)
Reported by: kfister
Patches:
16153-1.6.2.0-rc5.patch uploaded by kfister (license 912)
slin16.sip.patch.1 uploaded by malcolmd (license 924)
Tested by: kfister, malcolmd
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271261 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17501)
Reported by: fabled
Patches:
asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448)
Tested by: malcolmd, fabled, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271231 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16293)
Reported by: malcolmd
Patches:
g719.passthrough.patch.7 uploaded by malcolmd (license 924)
format_g719.c uploaded by malcolmd (license 924)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270940 f38db490-d61c-443f-a65b-d21fe96a405b
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Per Tilghman's request on IRC (#asterisk-bugs).
(closes issue #17506)
Reported by: brycebaril
Tested by: pabelanger, tilghman
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After the manager http auth changes, we forgot to remove the manual
sending of the file. Also, ast_http_send adds two \r\n to the header that
is passed to it, so a trailing \r\n is removed from the Content-type
header. It might be better to change ast_http_send, but I don't like changing
the behavior of an API function.
(closes issue #17239)
Reported by: cjacobsen
Patches:
patch2.diff uploaded by cjacobsen (license 1029)
Tested by: lathama, cjacobsen
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270552 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15757)
Reported by: Marquis
Patches:
distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
Tested by: Marquis, lmadsen, marcelloceschia
Review: https://reviewboard.asterisk.org/r/351/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270519 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon, 14 Jun 2010) | 14 lines
Properly play first file in sort list.
When using sort=alpha we would always skip the first file
in the list first time through. We now check for that
properly.
(closes issue #17470)
Reported by: pabelanger
Patches:
sort.aplha.patch uploaded by pabelanger (license 224)
Tested by: lmadsen
Review: https://reviewboard.asterisk.org/r/703/
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