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876 Commits

Author SHA1 Message Date
Kevin P. Fleming
7b31b6f43d Convert all release tags to Opsound music-on-hold.
For more details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/



git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.2.22-netsec@212958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-18 20:42:51 +00:00
Russell Bryant
f156100fe4 importing files for 1.2.22-netsec release
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.2.22-netsec@75467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-17 21:11:12 +00:00
Russell Bryant
512d3df8ca Creating tag for the release of asterisk-1.2.22-netsec
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.2.22-netsec@75464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-17 21:09:53 +00:00
Russell Bryant
40ef94cfe6 get branch up to date to include ASA fixes
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2007-07-17 21:09:39 +00:00
Russell Bryant
37b1321769 Creating tag for the release of asterisk-1.2.22-netsec
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Russell Bryant
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Kevin P. Fleming
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Joshua Colp
6480d28862 Merged revisions 41768,41827,41830,41880,41882,41989,42014,42054 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r41768 | file | 2006-09-01 18:49:07 -0400 (Fri, 01 Sep 2006) | 2 lines

Only wipe the redirected audio & video IP/port if it's specified, and trigger a reinvite.

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r41827 | bweschke | 2006-09-03 10:16:08 -0400 (Sun, 03 Sep 2006) | 3 lines

 Setting a retry of 0 is generally not a good idea and shouldn't be allowed. (#7574 - reported by regin)


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r41830 | bweschke | 2006-09-03 10:50:59 -0400 (Sun, 03 Sep 2006) | 3 lines

 Let's NOT spy on Zap/psuedo channels, mmmmmmmmk? 


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r41880 | bweschke | 2006-09-03 13:13:38 -0400 (Sun, 03 Sep 2006) | 3 lines

 Don't keep trying the same member in certain strategies when members of the queue are unavailable (#7278 - diLLec reported and patched) - This should have been patched here first and then merged into /trunk. My bad!


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r41882 | bweschke | 2006-09-03 13:38:22 -0400 (Sun, 03 Sep 2006) | 3 lines

 Make sure the forwarded channel inherits variables appropriately when we receive a call forward in the queue. (#7867 - raarts reported and patched)


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r41989 | oej | 2006-09-04 11:46:07 -0400 (Mon, 04 Sep 2006) | 2 lines

Don't kill the pvt before we have sent ACK on CANCEL (needs more testing before making a release)

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r42014 | qwell | 2006-09-05 12:27:46 -0400 (Tue, 05 Sep 2006) | 4 lines

Small typo in zapata.conf.sample

Reported by ppyy in 7881

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r42054 | file | 2006-09-05 16:02:48 -0400 (Tue, 05 Sep 2006) | 2 lines

Merge in last round of spy fixes. This should hopefully eliminate all the issues people have been seeing by distinctly separating what each component (core/spy) is responsible for. Core is responsible for adding a spy to a channel, feeding frames to the spy, removing the spy from a channel, and telling the spy to stop. Spy is responsible for reading frames in, and cleaning up after itself.

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Kevin P. Fleming
63566eac42 Merged revisions 19812 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r19812 | kpfleming | 2006-04-13 12:40:21 -0500 (Thu, 13 Apr 2006) | 2 lines

oops... let's not set a variable and then immediately overwrite it while assuming its old value will magically return

........


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Kevin P. Fleming
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2006-02-09 22:03:14 +00:00
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2006-02-09 03:01:45 +00:00
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2006-02-09 02:01:45 +00:00
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2006-02-08 23:00:44 +00:00
Kevin P. Fleming
f6d4da1706 simplify automerge setup
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2006-02-06 20:47:11 +00:00
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2006-02-05 18:00:23 +00:00
Kevin P. Fleming
4f44b40d33 update to current 1.2 branch
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2006-02-02 23:47:35 +00:00
Kevin P. Fleming
08e0a56f32 setup for new automerge process
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2006-02-02 23:45:00 +00:00
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2006-02-02 19:01:36 +00:00
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2006-02-02 17:01:37 +00:00
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2006-01-30 18:01:34 +00:00
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2006-01-28 14:01:17 +00:00
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2006-01-27 09:01:17 +00:00
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2006-01-27 01:01:17 +00:00
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6a02d8a8d9 automerge commit
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2006-01-26 20:01:20 +00:00
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2006-01-26 15:01:17 +00:00
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2006-01-25 20:01:00 +00:00
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2006-01-25 19:01:09 +00:00
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2006-01-25 10:01:13 +00:00
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2006-01-25 06:00:59 +00:00
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2006-01-25 02:01:00 +00:00
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2006-01-24 23:00:59 +00:00
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2e10d5244e automerge commit
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2006-01-24 21:01:03 +00:00
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2006-01-24 20:00:59 +00:00
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2006-01-24 14:01:00 +00:00
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2006-01-22 20:00:19 +00:00
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30c9cea332 automerge commit
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2006-01-22 18:00:19 +00:00
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2006-01-22 16:00:19 +00:00
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2006-01-22 09:00:21 +00:00
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2006-01-22 03:00:19 +00:00
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2006-01-22 00:00:10 +00:00
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2006-01-21 19:00:20 +00:00
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abcbea02c1 automerge commit
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2006-01-20 19:01:00 +00:00
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65850c0467 automerge commit
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2006-01-20 01:00:57 +00:00
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413b215b6c automerge commit
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2006-01-19 20:00:55 +00:00
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30185fdff4 automerge commit
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2006-01-19 05:00:39 +00:00
Kevin P. Fleming
8cedcec2bc add README note to the proper place for future releases
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2006-01-18 23:14:04 +00:00
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1cc6f6d3bb automerge commit
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2006-01-18 22:00:37 +00:00
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ca0a371996 automerge commit
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2006-01-18 03:00:42 +00:00
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b0fd9f9b74 automerge commit
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2006-01-18 02:00:34 +00:00
Kevin P. Fleming
396f6561a4 set up for automerge
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2006-01-17 23:58:15 +00:00
Kevin P. Fleming
f16619507a updating to current 1.2
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2006-01-13 03:52:45 +00:00
Kevin P. Fleming
69ac155838 initial import of Asterisk SIP support for network security devices
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2006-01-10 03:10:34 +00:00
Kevin P. Fleming
a6bae85952 prepare for merging changes from main 1.2 branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@7926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-10 03:05:18 +00:00
Kevin P. Fleming
494b9c486e create branch to hold network security device code
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@7925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-10 03:03:29 +00:00
Kevin P. Fleming
d8c5d338cb re-initialize _all_ sequence numbers when transfer completes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-09 22:48:48 +00:00
Russell Bryant
417438fea0 add missing unlock (issue #6112)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-09 22:07:26 +00:00
Tilghman Lesher
5488ffb6d4 Bug 6157 - Memory leak
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-09 20:08:24 +00:00
Tilghman Lesher
e8725ce940 Update variable documentation to match the code
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-09 18:37:50 +00:00
Kevin P. Fleming
0e734920e8 commit user/group-related changes from trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-09 18:11:23 +00:00
Kevin P. Fleming
ad598a2060 backport fix from revision 7856 of trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-09 18:09:53 +00:00
Kevin P. Fleming
bb47f95262 fix breakage introduced in revision 7871
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-09 18:08:07 +00:00
Russell Bryant
a7a2115b9c fix seg fault when using greek syntax in VoicemMailMain (issue #6142)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-09 05:11:44 +00:00
Russell Bryant
18429f2a69 backport fix for unnecessary unlock (issue #6171)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-09 04:52:16 +00:00
Tilghman Lesher
7e6430787c Bug 6156 - catch all threading errors, not just simple failure
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-07 07:27:18 +00:00
Tilghman Lesher
dc3ba8c223 Dumb error messages - "Context 'context' already included in 'in' context"
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-06 00:34:40 +00:00
Kevin P. Fleming
2fd6520a7e update agent persistence when an agent gets logged off by autologoff
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-06 00:21:00 +00:00
Tilghman Lesher
7a61b36104 Bug 6076 - Fix documentation of ast_trim_blank return value
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-05 23:53:07 +00:00
Kevin P. Fleming
2f4783a770 eliminate rounding errors that caused call time limits to be inaccurate (issue #5913)
round 'time left' reported during call limit warnings up to sound more accurate


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-05 23:49:50 +00:00
Tilghman Lesher
6056391a78 Bug 6081 - fix for memory leak, formatting fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-05 23:07:08 +00:00
Kevin P. Fleming
213b3575ce ensure that variable is initialized
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-05 20:52:38 +00:00
Olle Johansson
3821e18d08 Fix copyright of changed file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-05 09:13:21 +00:00
Kevin P. Fleming
224c6e3f7c send device state updates for auto-logoff of agents as well
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-05 00:58:29 +00:00
Kevin P. Fleming
18ab6f429c doh... fseek() has no useful return value
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2006-01-05 00:18:46 +00:00
Kevin P. Fleming
07f8542ce7 use proper fwrite() parameters and return value
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2006-01-04 23:51:03 +00:00
Kevin P. Fleming
cf293c224d return properly after extending file
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2006-01-04 23:45:34 +00:00
Kevin P. Fleming
e234ad0011 ensure that ulaw/alaw sound files are filled with silence when extended (not zeroes)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-04 23:27:57 +00:00
Kevin P. Fleming
229b778b0b make monitoring more tolerant of peers that deliver frames in bursts
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2006-01-04 23:02:38 +00:00
Olle Johansson
66f484defe Issue #5980: Removing extra CR+LF in manager events - needs port to trunk
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2006-01-04 21:46:40 +00:00
Olle Johansson
ed523e248a Fixing typo in XML for video updates.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-04 21:43:14 +00:00
Russell Bryant
3c49f27bf4 use a more correct way of determining the size of the destination buffer
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-04 07:06:50 +00:00
BJ Weschke
226d146b25 Fix the 'if' clause to be true under the right conditions. Bug #6126
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-01-04 05:27:38 +00:00
Kevin P. Fleming
1892289329 remove unused 'old' expression parser
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2006-01-03 20:22:18 +00:00
Tilghman Lesher
5529fc0355 Bug 6121 - typo in application description
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2006-01-03 18:15:12 +00:00
Kevin P. Fleming
73ef559240 revert incorrect fix for bug #6048 from revision 7709
put in correct (simpler) fix
add doxygen docs for channel spy 'state' values


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2006-01-03 17:24:56 +00:00
Kevin P. Fleming
b59a63c8cd backport rport scanning fix from trunk (bug #6071)
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2006-01-03 17:00:01 +00:00
Kevin P. Fleming
c36927137f don't leak memory for (most) expression evaluations
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2006-01-03 16:34:12 +00:00
Tilghman Lesher
649aaad100 Bug 6084 - MixMonitor after a 'cli stop monitor' deadlocks
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2006-01-02 07:31:54 +00:00
BJ Weschke
ea4c262482 Fix compiler warnings.
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2006-01-02 02:04:14 +00:00
Tilghman Lesher
89de6c9df6 Bug 6091 - Fix race condition around uniqueid
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2005-12-30 14:54:19 +00:00
Russell Bryant
4578bc3b0c fix memory leak in build_rpid (issue #6070)
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2005-12-28 17:35:56 +00:00
Russell Bryant
8df1e4866a backport fix for permissions of created recordings (issue #6067)
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2005-12-27 21:07:08 +00:00
Kevin P. Fleming
9f92220a9c backport fix to ensure that DSP is never enabled on pseudo channels
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-27 00:07:45 +00:00
Tilghman Lesher
09a9e7bcb7 Remove copy of code in libc, preferring code in utils.c (public domain code)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-26 20:32:50 +00:00
Russell Bryant
a434f8877a cast time_t to an int in printf/scanf (issue #5635)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-26 18:19:12 +00:00
Tilghman Lesher
ef891dc0a7 Bug 6051 - VMCOUNT should work as documented and count all, not quit after finding 1
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-23 06:38:08 +00:00
Kevin P. Fleming
3ebe29c983 add license reference to copyright notice displayed when CLI session begins
add 'show warranty' and 'show license' CLI commands (still need a complete list of non-GPL components included in Asterisk)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-23 03:01:24 +00:00
BJ Weschke
1028e43464 Another app documentation tweak.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-23 00:00:11 +00:00
Tilghman Lesher
4979f66f28 Bug 6050 SQL requires the use of single ticks to delimit values, not quotes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-22 22:04:13 +00:00
Russell Bryant
a5e9fe97c8 revert changes to videosupport to allow per-peer setting, since it isn't quite
complete and there is not an obvious fix at this point


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-22 20:36:47 +00:00
Russell Bryant
2293cfaf84 remove stray unlock (issue #5955)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-22 16:17:43 +00:00
Josh Roberson
d252dde933 Actually put in the per-peer settings for sip video, as they didn't make it in at astricon somehow, and I've been too busy up until now to redo it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-21 22:23:39 +00:00
Tilghman Lesher
37a49a5992 Allow a chan_alsa that failed to open sound devices to be unloaded.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-21 20:01:16 +00:00
BJ Weschke
2783801610 Bug #6040 - Documentation correction
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-21 19:53:49 +00:00
Tilghman Lesher
82f19c9a91 Bug 5777 - Remove parentheses on Goto in AEL, so that it parses correctly
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-21 19:23:12 +00:00
Russell Bryant
71d9531e45 check array bounds when parsing arguments to AGI (issue #5868)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-20 20:21:26 +00:00
Russell Bryant
3bbeaefa57 backport fix for reloading peer context (issue #6007)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-20 18:05:45 +00:00
Russell Bryant
4611be805a backport fix for segfault on directed pickup when no CDR is available (issue #5998)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-20 17:34:00 +00:00
Kevin P. Fleming
587493fdfc backport fix for larger-than-20ms-frames from trunk (bug #5697)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-20 12:58:37 +00:00
Russell Bryant
50079144e3 I messed up and accidently committed this to the trunk first ...
- add note on required values of sip_methods struct
- remove duplicate function prototype
- remove duplicate ast_mutex_lock (issue #6025)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-19 23:47:23 +00:00
Tilghman Lesher
2ca80e76a0 Bug 5988 - record append option not working
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-19 19:06:46 +00:00
Tilghman Lesher
68e6502484 Bug 6026 - segfault for the sequence NoCDR(), SetAMAFlags()
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-19 05:38:44 +00:00
Tilghman Lesher
1e91dfa5df Document that curley braces must be on the same line as the keyword.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-17 18:55:38 +00:00
Tilghman Lesher
e4e5f7c65b Bug 6009 - off by one error
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-17 17:19:32 +00:00
Kevin P. Fleming
8ef66261f1 Max-Forwards headers must only be present on requests, not responses
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7515 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-17 03:59:05 +00:00
Kevin P. Fleming
90812b536c forcibly expire previous subscriptions from a peer when they resubscribe (keeps them from building up and waiting for expiration, and stops us sending unwanted NOTIFY messages to devices)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-17 03:44:30 +00:00
Kevin P. Fleming
2b880d3ef8 fix some buglet when building team branch version strings
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-17 02:20:04 +00:00
Tilghman Lesher
6ab729aeb3 We want to check the previous value, not the current value (which was just changed).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-17 01:02:29 +00:00
Tilghman Lesher
c9f97f49c5 First field is truncated
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-16 00:49:39 +00:00
Christian Richter
8b352fcb94 * Added mISDN/mISDNuser Echo cancel Patch
* Fixed Makefiles so that chan_misdn can be compiled again
* added some hints, that mISDN cannot be compiled against gcc-4, SMP, Spinlock Debug
* fixed some Minor issues in chan_misdn, regarding Type Of Number and Presentation






git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-15 10:52:30 +00:00
BJ Weschke
5a06af4d57 Bug #6003 - Don't free the channel structure until after having sent the manager event.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-15 02:51:54 +00:00
Kevin P. Fleming
2ede900bfd clarify substring documentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-13 18:54:22 +00:00
Kevin P. Fleming
fb0ffc7784 correct broken math in tvfix() for timestamp values over one million
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-13 16:06:27 +00:00
Kevin P. Fleming
e69171dc06 restore ability of caller to hangup calls that are still ringing (issue #5839)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-13 05:53:00 +00:00
Kevin P. Fleming
8452f6b494 ensure that hangups while incoming calls are in early state are handled properly (issue #5919)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-13 05:14:27 +00:00
Kevin P. Fleming
f9bc6f8afc only report AGENT_IDLE for callback mode agents when they are actually idle (issue #5902)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-13 04:43:38 +00:00
Kevin P. Fleming
f9f44d7d9b use the stream's current point when pausing/unpausing, instead of elapsed time (which doesn't work when the stream has been skipped forward or backward) (issue #5897)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-13 04:25:14 +00:00
Kevin P. Fleming
86bb1472cc set all the child file descriptors to non-blocking so that we don't hang if the child fails to send a newline-terminated command or error message
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-12 17:30:59 +00:00
Tilghman Lesher
ef09c506c8 Typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-12 17:19:29 +00:00
Russell Bryant
af7b3b420e silence a couple of compiler warnings about pointer signedness
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-11 06:08:56 +00:00
Tilghman Lesher
3b80a35a66 Bug 5965 - major bug in AST_LIST_REMOVE
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-11 01:26:51 +00:00
Tilghman Lesher
f59b27258c Bug 5967
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-11 01:15:15 +00:00
BJ Weschke
d38fff38f9 Bug #5877
Make sure the digit string from E&M wink DNIS collection is properly null terminated as it grows.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-10 18:10:50 +00:00
Tilghman Lesher
91c5b12f25 Bug 5960
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-08 23:45:36 +00:00
Tilghman Lesher
894bdd5cda Documenting two keywords that were previously missing
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-08 23:32:08 +00:00
Kevin P. Fleming
d8c58ca6c8 initialize the buffer before using it...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-08 01:05:43 +00:00
Kevin P. Fleming
43f7956422 ensure that hints are allowed to use global variable references
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-07 20:46:55 +00:00
Kevin P. Fleming
96fcc02900 ensure that sound/music files are handled as binary
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-06 00:23:05 +00:00
Kevin P. Fleming
2e8b6e7f20 replace with unmodified file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-06 00:19:17 +00:00
Kevin P. Fleming
3ed93398e9 _really_ don't mess with line endings
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-06 00:08:19 +00:00
Kevin P. Fleming
d84e0caea8 don't try to expand keywords in this file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-06 00:07:48 +00:00
Kevin P. Fleming
5c1da58b90 set this file to binary mode, so it won't be mangled on export/checkout
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-06 00:02:08 +00:00
Russell Bryant
018c617199 remove ASTERISKVERSIONNUM from the version string given to doxygen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7340 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-05 06:47:51 +00:00
Russell Bryant
a49f954f6a don't delete dynamic queue members when reloading the static members from a realtime database (issue #5922)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-05 00:09:12 +00:00
Russell Bryant
03a8791440 fix the order of arguments to an error message (issue #5927)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-05 00:01:17 +00:00
Kevin P. Fleming
78ad17381e use a more efficient way to get the revision number, that will also report if the working copy contains uncommitted modifications
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-04 18:03:07 +00:00
Tilghman Lesher
5b03803e2f Bug 5925: check for "Unknown", as that's what app_voicemail puts into the field for Unknown callerid
Also, remove useless res checks (initialized to 0; never set)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-03 19:55:05 +00:00
Olle Johansson
ee1e461a57 Documenting the default registerattempts setting as 0, continue hammering the server for ever and ever ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-03 01:24:40 +00:00
Tilghman Lesher
a543725311 Turn on executable bits for startup scripts, and fix bash var interpolation for Mandrake
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-02 21:12:05 +00:00
Olle Johansson
8c701a0861 Bug #5907. Improve SIP INFO DTMF debugging output. (1.2 & Trunk)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-02 00:52:13 +00:00
Kevin P. Fleming
5fbe566524 inherit channel variables into channels created by Page() application (issue #5888)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-02 00:51:15 +00:00
Kevin P. Fleming
e14cf31bc9 allow previous context-searching behavior to be used if desired (issue #5899)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-02 00:42:40 +00:00
Kevin P. Fleming
0042c03c7e properly handle password changes when mailbox is last line of config file and not followed by a newline (issue #5870)
reformat password changing code to conform to coding guidelines (issue #5870)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-02 00:39:00 +00:00
Kevin P. Fleming
41023ecb9c protect agent_bridgedchannel() from segfaulting when there is no bridged channel (issue #5879)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-02 00:28:48 +00:00
Kevin P. Fleming
fb8d0a544d allow variables to exist on both 'halves' of the Local channel (issue #5810)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-02 00:26:12 +00:00
Kevin P. Fleming
acaf8c9cc9 don't block waiting for the Festival server forever when it goes away (issue #5882)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 23:49:44 +00:00
Kevin P. Fleming
0d3fc8d103 ensure channel's scheduling context is freed (issue #5788)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 23:34:58 +00:00
Kevin P. Fleming
9bbfbacacc Makefile 'update' target now supports updating from Subversion repositories (issue #5875)
remove support for 'patches' subdirectory, it's no longer useful


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 23:18:29 +00:00
Olle Johansson
8e301c629f Changing bug report address to the Asterisk issue tracker
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 23:18:14 +00:00
Olle Johansson
9576d8a081 Removing references to 1.1dev, replacing with 1.2, in documentation files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 22:21:13 +00:00
Olle Johansson
7db96672bf Fixing some spelling errors, as well as changing "cvs" to "subversion" in misdn documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 21:38:38 +00:00
Kevin P. Fleming
0d8f099ca2 ensure that 'svn info' output is in the expected language for the script to parse (issue #5880)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 19:25:06 +00:00
Russell Bryant
0a4d7e5068 use ast_app_separate_args to split arguments (issue #5686)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 02:33:58 +00:00
Russell Bryant
a32f75b800 fix queue weight feature - compare member interfaces instead of pointers to
the members, since each queue has its own list of members. (issue #5863)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 01:19:04 +00:00
Russell Bryant
78f37c51ef use '=' instead of '==' for string comparisons. /bin/bash is ok with this, but
/bin/sh is not.  (issue #5885)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-12-01 00:44:34 +00:00
Russell Bryant
d6e5eb75ee remove outdated redhat init script and provide the updated one in 'make rpm' (issue #5786)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-30 21:31:18 +00:00
Russell Bryant
6f433a7e7b Comment out LD_ASSUME_KERNEL by default.
Print error messages if the asterisk executable or the asterisk configuration
directory are not found. (issue #5785, #5708)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-30 21:24:52 +00:00
Russell Bryant
ce2ecb5582 fix DIALEDTIME when call has not been answered (issue #5862)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-30 17:39:36 +00:00
Russell Bryant
45c15e8561 do not allow an rtp message with zero type (issue #5749)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-30 14:27:59 +00:00
Russell Bryant
5e3f4186da fix hint case sensitivity (issue #5856)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-30 07:22:42 +00:00
Russell Bryant
d03694df19 add description of the "fromdomain" option (issue #5874)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-30 05:26:26 +00:00
Josh Roberson
d52e8fa7d2 backport fix from trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-30 03:52:18 +00:00
Kevin P. Fleming
77c6cd0bff remove remaining CVS references
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-30 03:37:37 +00:00
Kevin P. Fleming
463ebe8b4b port memory leak fix from rev 7223 in trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-30 03:20:42 +00:00
Kevin P. Fleming
a563eab49a remove extraneous svn:executable properties
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29 18:24:39 +00:00
Kevin P. Fleming
bf9d4b103d do the multiple-lock check for cond_wait properly...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29 14:26:22 +00:00
Russell Bryant
7dc2449f3a print an error message if invalid arguments are specified
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29 06:12:29 +00:00
Russell Bryant
27cc20db11 fix a couple of typos and a buglet
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29 05:11:05 +00:00
Kevin P. Fleming
4e04f5f731 if the lock protected a pthread_cond is held recursively, warn before waiting onthe condition
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29 01:25:28 +00:00
Kevin P. Fleming
5ec3844077 remove CVS ignore lists, set SVN ignore lists
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29 00:46:54 +00:00
Kevin P. Fleming
f641c7cd0d port version string computation from trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-28 04:01:06 +00:00
Kevin P. Fleming
49aa54cafc branch renames
remove unneeded branches


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@7199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-27 16:04:52 +00:00
Russell Bryant
e3b81e66a1 issue #5850
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-25 14:17:38 +00:00
Russell Bryant
7bb6b928b5 fix typo - thanks twisted :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7194 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-23 06:02:32 +00:00
Russell Bryant
ad4dc7c4a9 fix the date
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-23 05:59:05 +00:00
Russell Bryant
feb9764721 issue #5828
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-23 05:57:10 +00:00
Russell Bryant
8c74501001 issue #5826
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-23 05:49:35 +00:00
Russell Bryant
5861d3fd64 issue #5829
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7189 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-23 05:43:27 +00:00
Kevin P. Fleming
08acdc6a86 backport fix from HEAD branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 19:34:09 +00:00
Kevin P. Fleming
3b50570c9b backport fix from HEAD branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 19:31:26 +00:00
Kevin P. Fleming
00d7e962a7 backport fix from HEAD branch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 19:29:51 +00:00
Russell Bryant
192f7413c2 revert previous change for Darwin
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 15:57:51 +00:00
Russell Bryant
5bdb8074aa issue #5789
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 13:40:34 +00:00
Russell Bryant
6abe5e1b23 don't hardcode poll.o for Darwin
fix incorrect portion of yesterday's Solaris fixes


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 13:23:53 +00:00
Russell Bryant
4612b4ccca issue #5815
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 13:17:32 +00:00
Russell Bryant
3245bf61a1 issue #5775
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 02:44:06 +00:00
Russell Bryant
9ee13167bb issue #5787
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 02:03:23 +00:00
Russell Bryant
ca9633ca9b issue #5794
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 01:00:51 +00:00
Russell Bryant
e04884545c issue #5806
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-21 00:03:40 +00:00
Russell Bryant
aa2239a050 issue #5773
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-20 23:56:49 +00:00
Russell Bryant
40a1b60cbd issue #5774
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-20 23:51:58 +00:00
Russell Bryant
cccb11916f fix logic for n+101 jumps
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-20 23:42:43 +00:00
Russell Bryant
49641825eb issue #5795
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-20 23:36:45 +00:00
Russell Bryant
c14c078405 issue #5792
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-20 23:33:03 +00:00
Russell Bryant
f2af073588 issue #5799
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-20 23:27:28 +00:00
Russell Bryant
b9a5eddb6d fix issue number for previous commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-20 23:22:10 +00:00
Russell Bryant
bffc815375 issue #5791
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-20 23:21:40 +00:00
Russell Bryant
5c3f322364 Shorten the module description (issue #5791)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-20 23:18:01 +00:00
Russell Bryant
3a6fae817c fix the output of Makefile generated variables to doxygen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-18 04:13:39 +00:00
Russell Bryant
d8f74ecb94 Add missing carriage return and line feed to the SDP line indicating that we don't support VAD (issue #5780)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-17 20:26:27 +00:00
Kevin P. Fleming
f7f2b1b033 fix remaining breakage related to chan_modem modules
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-17 04:26:07 +00:00
Admin Commit
601ab7ff82 This commit was manufactured by cvs2svn to create branch 'v1-2'.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-2@7126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-16 22:40:43 +00:00
1165 changed files with 25405 additions and 11832 deletions

2
.cleancount Executable file → Normal file
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@@ -1 +1 @@
8
9

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@@ -1,16 +0,0 @@
asterisk
defaults.h
ast_expr2.output
.version
.depend
.applied
mpg123-0.59r
mpg123-0.59r.tar.gz
update.out
.lastclean
.cleancount
.tags-depend
.tags-sources
tags
TAGS
testexpr2

1
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@@ -0,0 +1 @@
9

1
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@@ -0,0 +1 @@
1.2.22-netsec

2
BUGS Executable file → Normal file
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@@ -10,7 +10,7 @@ For more information on using the bug tracker, or to
learn how you can contribute by acting as a bug marshall
please see:
http://www.digium.com/index.php?menu=bugguidelines
http://www.asterisk.org/developers/bug-guidelines
If you would like to submit a feature request, please
resist the temptation to post it to the bug tracker.

0
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0
COPYING Executable file → Normal file
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12
CREDITS Executable file → Normal file
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@@ -13,6 +13,8 @@ Telesthetic - for supporting SIP development
Christos Ricudis - for substantial code contributions
nic.at - ENUM support in Asterisk
Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development
=== WISHLIST CONTRIBUTERS ===
@@ -31,6 +33,14 @@ Jim Dixon - Zapata Telephony and app_rpt
Russell Bryant - Asterisk 1.0 maintainer and misc. enhancements
russelb@clemson.edu
Anthony Minessale II - Countless big and small fixes, and relentless forward push
ChanSpy, ForkCDR, ControlPlayback, While/EndWhile, DumpChan, Dictate,
MacroIf, ExecIf, ExecIfTime, RetryDial, MixMonitor applications; res_odbc;
many realtime concepts and implementation pieces, including res_config_odbc;
format_slin; cdr_custom; several features in Dial including L(), G() and
enhancements to M() and D(); several CDR enhancements including CDR variables;
attended transfer; one touch record; native MOH; manager eventmask; command
line '-t' flag to allow recording/voicemail on nfs shares; #exec command and
multiline comments in config files; setvar in iax and sip configs.
anthmct@yahoo.com http://www.asterlink.com
James Golovich - Innumerable contributions
You can find him and asterisk-perl at http://asterisk.gnuinter.net
@@ -85,7 +95,7 @@ Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
available under a Creative Commons License at http://www.asteriskdocs.org
=== HOLD MUSIC ===
Music provided by www.freeplaymusic.com
Music provided by www.opsound.org
=== OTHER SOURCE CODE IN ASTERISK ===

6008
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0
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12
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@@ -5,6 +5,18 @@ applies to all loadable Asterisk modules used on your system as well,
except as defined below. The GPL (version 2) is included in this
source tree in the file COPYING.
This package also includes various components that are not part of
Asterisk itself; these components are in the 'contrib' directory
and its subdirectories. Most of these components are also
distributed under the GPL version 2 as well, except for the following:
contrib/firmware/iax/iaxy.bin:
This file is Copyright (C) Digium, Inc. and is licensed for
use with Digium IAXy hardware devices only. It can be
distributed freely as long as the distribution is in the
original form present in this package (not reformatted or
modified).
Digium, Inc. (formerly Linux Support Services) holds copyright
and/or sufficient licenses to all components of the Asterisk
package, and therefore can grant, at its sole discretion, the ability

225
Makefile Executable file → Normal file
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@@ -28,6 +28,7 @@ HOST_CC=gcc
ifeq ($(CROSS_COMPILE),)
OSARCH=$(shell uname -s)
OSREV=$(shell uname -r)
OSCPU=$(shell uname -m)
else
OSARCH=$(CROSS_ARCH)
OSREV=$(CROSS_REV)
@@ -36,6 +37,8 @@ endif
# Remember the MAKELEVEL at the top
MAKETOPLEVEL?=$(MAKELEVEL)
PWD=$(shell pwd)
ifneq ($(findstring dont-optimize,$(MAKECMDGOALS)),dont-optimize)
######### More GSM codec optimization
######### Uncomment to enable MMXTM optimizations for x86 architecture CPU's
@@ -81,6 +84,8 @@ TRACE_FRAMES = #-DTRACE_FRAMES
#
MALLOC_DEBUG = #-include $(PWD)/include/asterisk/astmm.h
INSTALL=install
# Where to install asterisk after compiling
# Default -> leave empty
INSTALL_PREFIX?=
@@ -105,6 +110,9 @@ BUSYDETECT+= #-DBUSYDETECT_TONEONLY
# Don't use together with -DBUSYDETECT_TONEONLY
BUSYDETECT+= #-DBUSYDETECT_COMPARE_TONE_AND_SILENCE
# Comment this if you want to disable MIDCOM
MIDCOM = -DMIDCOM
ifneq ($(OSARCH),SunOS)
ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk
ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/asterisk
@@ -179,7 +187,6 @@ ifeq ($(OSARCH),Linux)
# You must have GCC 3.4 to use k8, otherwise use athlon
PROC=k8
#PROC=athlon
OPTIONS+=-m64
endif
ifeq ($(PROC),sparc64)
@@ -200,21 +207,25 @@ ifeq ($(OSARCH),Linux)
OPTIONS+=-fsigned-char -mcpu=ep9312
else
ifeq ($(SUB_PROC),xscale)
OPTIONS+=-fsigned-char -msoft-float -mcpu=xscale
OPTIONS+=-fsigned-char -mcpu=xscale
else
OPTIONS+=-fsigned-char -msoft-float
OPTIONS+=-fsigned-char
endif
endif
endif
MPG123TARG=linux
endif
PWD=$(shell pwd)
GREP=grep
LN=ln
ID=id
ifeq ($(OSARCH),SunOS)
GREP=/usr/xpg4/bin/grep
M4=/usr/local/bin/m4
ID=/usr/xpg4/bin/id
LN=/usr/xpg4/bin/ln
INSTALL=ginstall
endif
INCLUDE+=-Iinclude -I../include
@@ -281,7 +292,10 @@ endif
ifeq ($(OSARCH),SunOS)
ASTCFLAGS+=-Wcast-align -DSOLARIS
INCLUDE+=-Iinclude/solaris-compat -I$(CROSS_COMPILE_TARGET)/usr/local/ssl/include
INCLUDE+=-Iinclude/solaris-compat -I$(CROSS_COMPILE_TARGET)/opt/ssl/include -I$(CROSS_COMPILE_TARGET)/usr/local/ssl/include
ifeq ($(OSCPU),sun4u)
OPTIMIZE+=-mcpu=v9 -mcpu=ultrasparc
endif
endif
ifeq ($(findstring CYGWIN,$(OSARCH)),CYGWIN)
@@ -305,30 +319,35 @@ endif # WITHOUT_ZAPTEL
LIBEDIT=editline/libedit.a
ifneq ($(wildcard .version),)
ASTERISKVERSION=$(shell cat .version)
ASTERISKVERSIONNUM=$(shell awk -F. '{printf "%02d%02d%02d", $$1, $$2, $$3}' .version)
RPMVERSION=$(shell sed 's/[-\/:]/_/g' .version)
ASTERISKVERSION:=$(shell cat .version)
ASTERISKVERSIONNUM:=$(shell awk -F. '{printf "%01d%02d%02d", $$1, $$2, $$3}' .version)
RPMVERSION:=$(shell sed 's/[-\/:]/_/g' .version)
else
RPMVERSION=unknown
endif
ifneq ($(wildcard CVS),)
# CVS mirrors of SVN have .svnrevision files showing
# which SVN revision they are based on, and .svnbranch
# showing the branch they are made from
ifneq ($(wildcard .svnrevision),)
ASTERISKVERSIONNUM=999999
ifneq ($(wildcard CVS/Tag),)
ASTERISKVERSION=$(shell echo "CVS-`sed 's/^T//g' CVS/Tag`-`date +"%D-%T"`")
else
ASTERISKVERSION=CVS HEAD
endif
ASTERISKVERSION:=SVN-$(shell cat .svnbranch)-r$(shell cat .svnrevision)
else
ASTERISKVERSIONNUM=000000
ifneq ($(wildcard .svn),)
ASTERISKVERSIONNUM=999999
ASTERISKVERSION=SVN-$(shell build_tools/make_svn_branch_name)
endif
endif
ASTCFLAGS+= $(DEBUG_THREADS)
ASTCFLAGS+= $(TRACE_FRAMES)
ASTCFLAGS+= $(MALLOC_DEBUG)
ASTCFLAGS+= $(BUSYDETECT)
ASTCFLAGS+= $(MIDCOM)
ASTCFLAGS+= $(OPTIONS)
ifneq ($(findstring dont-optimize,$(MAKECMDGOALS)),dont-optimize)
ASTCFLAGS+= -fomit-frame-pointer
endif
SUBDIRS=res channels pbx apps codecs formats agi cdr funcs utils stdtime
OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o \
@@ -354,7 +373,11 @@ endif
ifeq ($(OSARCH),Linux)
LIBS+=-ldl -lpthread -lncurses -lm -lresolv #-lnjamd
else
LIBS+=-lncurses -lm
ifeq ($(OSARCH),SunOS)
LIBS+=-lm -lcurses
else
LIBS+=-lncurses -lm
endif
endif
ifeq ($(OSARCH),Darwin)
@@ -363,7 +386,7 @@ ifeq ($(OSARCH),Darwin)
AUDIO_LIBS=-framework CoreAudio
ASTLINK=-Wl,-dynamic
SOLINK=-dynamic -bundle -undefined suppress -force_flat_namespace
OBJS+=poll.o
OBJS+= poll.o
ASTCFLAGS+=-DPOLLCOMPAT
else
#These are used for all but Darwin
@@ -384,10 +407,10 @@ ifeq ($(OSARCH),OpenBSD)
endif
ifeq ($(OSARCH),SunOS)
LIBS+=-lpthread -ldl -lnsl -lsocket -lresolv -L$(CROSS_COMPILE_TARGET)/usr/local/ssl/lib
LIBS+=-lpthread -ldl -lnsl -lsocket -lresolv -L$(CROSS_COMPILE_TARGET)/opt/ssl/lib -L$(CROSS_COMPILE_TARGET)/usr/local/ssl/lib
OBJS+=strcompat.o
ASTLINK=
SOLINK=-shared -fpic -L$(CROSS_COMPILE_TARGET)/usr/local/ssl/lib
SOLINK=-shared -fpic -L$(CROSS_COMPILE_TARGET)/opt/ssl/lib -L$(CROSS_COMPILE_TARGET)/usr/local/ssl/lib
endif
ifeq ($(MAKETOPLEVEL),$(MAKELEVEL))
@@ -403,8 +426,6 @@ endif
LIBS+=-lssl
INSTALL=install
_all: all
@echo " +--------- Asterisk Build Complete ---------+"
@echo " + Asterisk has successfully been built, but +"
@@ -427,10 +448,10 @@ endif
noclean: depend asterisk subdirs
editline/config.h:
cd editline && unset CFLAGS LIBS && ./configure ; \
cd editline && unset CFLAGS LIBS && CFLAGS="$(OPTIMIZE)" ./configure ; \
editline/libedit.a: FORCE
cd editline && unset CFLAGS LIBS && test -f config.h || ./configure
cd editline && unset CFLAGS LIBS && test -f config.h || CFLAGS="$(OPTIMIZE)" ./configure
$(MAKE) -C editline libedit.a
db1-ast/libdb1.a: FORCE
@@ -534,8 +555,9 @@ clean:
$(MAKE) -C stdtime clean
datafiles: all
if test $$(id -u) = 0; then sh mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
if [ x`$(ID) -un` = xroot ]; then sh mkpkgconfig $(DESTDIR)/usr/lib/pkgconfig; fi
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/digits
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/silence
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/priv-callerintros
for x in sounds/digits/*.gsm; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
@@ -545,6 +567,14 @@ datafiles: all
exit 1; \
fi; \
done
for x in sounds/silence/*.gsm; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds/silence ; \
else \
echo "No description for $$x"; \
exit 1; \
fi; \
done
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds/dictate
for x in sounds/dictate/*.gsm; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
@@ -572,7 +602,7 @@ datafiles: all
exit 1; \
fi; \
done
for x in sounds/demo-* sounds/vm-* sounds/transfer* sounds/pbx-* sounds/ss-* sounds/beep* sounds/dir-* sounds/conf-* sounds/agent-* sounds/invalid* sounds/tt-* sounds/auth-* sounds/privacy-* sounds/queue-* sounds/spy-* sounds/priv-* sounds/screen-* sounds/hello-*; do \
for x in sounds/demo-* sounds/vm-* sounds/transfer* sounds/pbx-* sounds/ss-* sounds/beep* sounds/dir-* sounds/conf-* sounds/agent-* sounds/invalid* sounds/tt-* sounds/auth-* sounds/privacy-* sounds/queue-* sounds/spy-* sounds/priv-* sounds/screen-* sounds/hello-* sounds/hours* sounds/minute* sounds/second* ; do \
if $(GREP) -q "^%`basename $$x`%" sounds.txt; then \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/sounds ; \
else \
@@ -588,17 +618,16 @@ datafiles: all
mkdir -p $(DESTDIR)$(AGI_DIR)
update:
@if [ -d CVS ]; then \
if [ -f patches/.applied ]; then \
patches=`cat patches/.applied`; \
fi; \
if [ ! -z "$$patches" ]; then \
for x in $$patches; do \
echo "Unapplying $$x..."; \
patch -R -p0 < patches/$$x; \
done; \
rm -f patches/.applied; \
@if [ -d .svn ]; then \
echo "Updating from Subversion..." ; \
svn update | tee update.out; \
rm -f .version; \
if [ `grep -c ^C update.out` -gt 0 ]; then \
echo ; echo "The following files have conflicts:" ; \
grep ^C update.out | cut -b4- ; \
fi ; \
rm -f update.out; \
elif [ -d CVS ]; then \
echo "Updating from CVS..." ; \
cvs -q -z3 update -Pd | tee update.out; \
rm -f .version; \
@@ -607,19 +636,8 @@ update:
grep ^C update.out | cut -d' ' -f2- ; \
fi ; \
rm -f update.out; \
if [ ! -z "$$patches" ]; then \
for x in $$patches; do \
if [ -f patches/$$x ]; then \
echo "Applying patch $$x..."; \
patch -p0 < patches/$$x; \
echo $$x >> patches/.applied; \
else \
echo "Patch $$x no longer relevant"; \
fi; \
done; \
fi; \
else \
echo "Not CVS"; \
echo "Not under version control"; \
fi
NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
@@ -636,23 +654,22 @@ bininstall: all
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/system
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/tmp
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/meetme
mkdir -p $(DESTDIR)$(ASTSPOOLDIR)/monitor
if [ -f asterisk ]; then $(INSTALL) -m 755 asterisk $(DESTDIR)$(ASTSBINDIR)/; fi
if [ -f cygwin/asterisk.exe ]; then $(INSTALL) -m 755 cygwin/asterisk.exe $(DESTDIR)$(ASTSBINDIR)/; fi
if [ -f asterisk.dll ]; then $(INSTALL) -m 755 asterisk.dll $(DESTDIR)$(ASTSBINDIR)/; fi
ln -sf asterisk $(DESTDIR)$(ASTSBINDIR)/rasterisk
$(LN) -sf asterisk $(DESTDIR)$(ASTSBINDIR)/rasterisk
$(INSTALL) -m 755 contrib/scripts/astgenkey $(DESTDIR)$(ASTSBINDIR)/
$(INSTALL) -m 755 contrib/scripts/autosupport $(DESTDIR)$(ASTSBINDIR)/
if [ ! -f $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ]; then \
cat contrib/scripts/safe_asterisk | sed 's|__ASTERISK_SBIN_DIR__|$(ASTSBINDIR)|;' > $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ;\
cat contrib/scripts/safe_asterisk | sed 's|__ASTERISK_SBIN_DIR__|$(ASTSBINDIR)|;s|__ASTERISK_VARRUN_DIR__|$(ASTVARRUNDIR)|;' > $(DESTDIR)$(ASTSBINDIR)/safe_asterisk ;\
chmod 755 $(DESTDIR)$(ASTSBINDIR)/safe_asterisk;\
fi
for x in $(SUBDIRS); do $(MAKE) -C $$x install || exit 1 ; done
$(INSTALL) -d $(DESTDIR)$(ASTHEADERDIR)
$(INSTALL) -m 644 include/asterisk/*.h $(DESTDIR)$(ASTHEADERDIR)
if [ -n "$(OLDHEADERS)" ]; then \
rm -f $(addprefix $(DESTDIR)$(ASTHEADERDIR)/,$(OLDHEADERS)) ;\
fi
rm -f $(DESTDIR)$(ASTVARLIBDIR)/sounds/voicemail
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/sounds
mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-csv
mkdir -p $(DESTDIR)$(ASTLOGDIR)/cdr-custom
@@ -671,30 +688,10 @@ bininstall: all
else \
echo "You need to do cvs update -d not just cvs update" ; \
fi
( cd $(DESTDIR)$(ASTVARLIBDIR)/sounds ; ln -s $(ASTSPOOLDIR)/voicemail . )
if [ -f mpg123-0.59r/mpg123 ]; then $(MAKE) -C mpg123-0.59r install; fi
@echo " +---- Asterisk Installation Complete -------+"
@echo " + +"
@echo " + YOU MUST READ THE SECURITY DOCUMENT +"
@echo " + +"
@echo " + Asterisk has successfully been installed. +"
@echo " + If you would like to install the sample +"
@echo " + configuration files (overwriting any +"
@echo " + existing config files), run: +"
@echo " + +"
@echo " + $(MAKE) samples +"
@echo " + +"
@echo " +----------------- or ---------------------+"
@echo " + +"
@echo " + You can go ahead and install the asterisk +"
@echo " + program documentation now or later run: +"
@echo " + +"
@echo " + $(MAKE) progdocs +"
@echo " + +"
@echo " + **Note** This requires that you have +"
@echo " + doxygen installed on your local system +"
@echo " +-------------------------------------------+"
@$(MAKE) -s oldmodcheck
install-subdirs:
for x in $(SUBDIRS); do $(MAKE) -C $$x install || exit 1 ; done
NEWMODS=$(notdir $(wildcard */*.so))
OLDMODS=$(filter-out $(NEWMODS),$(notdir $(wildcard $(DESTDIR)$(MODULES_DIR)/*.so)))
@@ -717,10 +714,32 @@ oldmodcheck:
echo " WARNING WARNING WARNING" ;\
fi
install: all datafiles bininstall
install: all datafiles bininstall install-subdirs
@if [ -x /usr/sbin/asterisk-post-install ]; then \
/usr/sbin/asterisk-post-install $(DESTDIR) . ; \
fi
@echo " +---- Asterisk Installation Complete -------+"
@echo " + +"
@echo " + YOU MUST READ THE SECURITY DOCUMENT +"
@echo " + +"
@echo " + Asterisk has successfully been installed. +"
@echo " + If you would like to install the sample +"
@echo " + configuration files (overwriting any +"
@echo " + existing config files), run: +"
@echo " + +"
@echo " + $(MAKE) samples +"
@echo " + +"
@echo " +----------------- or ---------------------+"
@echo " + +"
@echo " + You can go ahead and install the asterisk +"
@echo " + program documentation now or later run: +"
@echo " + +"
@echo " + $(MAKE) progdocs +"
@echo " + +"
@echo " + **Note** This requires that you have +"
@echo " + doxygen installed on your local system +"
@echo " +-------------------------------------------+"
@$(MAKE) -s oldmodcheck
upgrade: all bininstall
@@ -780,7 +799,7 @@ samples: adsi
fi; \
done
mkdir -p $(DESTDIR)$(ASTVARLIBDIR)/mohmp3 ; \
for x in sounds/*.mp3; do \
for x in sounds/moh/*.mp3 sounds/moh/LICENSE; do \
$(INSTALL) -m 644 $$x $(DESTDIR)$(ASTVARLIBDIR)/mohmp3 ; \
done
rm -f $(DESTDIR)$(ASTVARLIBDIR)/mohmp3/sample-hold.mp3
@@ -831,12 +850,12 @@ __rpm: include/asterisk/version.h spec
$(MAKE) DESTDIR=/tmp/asterisk install ; \
$(MAKE) DESTDIR=/tmp/asterisk samples ; \
mkdir -p /tmp/asterisk/etc/rc.d/init.d ; \
cp -f redhat/asterisk /tmp/asterisk/etc/rc.d/init.d/ ; \
cp -f contrib/init.d/rc.redhat.asterisk /tmp/asterisk/etc/rc.d/init.d/asterisk ; \
rpmbuild --rcfile /usr/lib/rpm/rpmrc:redhat/rpmrc -bb asterisk.spec
progdocs:
(cat contrib/asterisk-ng-doxygen; echo "HAVE_DOT=$(HAVEDOT) \
PROJECT_NUMBER=$(ASTERISKVERSION) - $(ASTERISKVERSIONNUM)") | doxygen -
(cat contrib/asterisk-ng-doxygen; echo "HAVE_DOT=$(HAVEDOT)"; \
echo "PROJECT_NUMBER=$(ASTERISKVERSION)") | doxygen -
mpg123:
@wget -V >/dev/null || (echo "You need wget" ; false )
@@ -849,7 +868,9 @@ config:
$(INSTALL) -m 755 contrib/init.d/rc.redhat.asterisk /etc/rc.d/init.d/asterisk; \
/sbin/chkconfig --add asterisk; \
elif [ -d /etc/init.d ]; then \
$(INSTALL) -m 755 init.asterisk /etc/init.d/asterisk; \
echo "Only distros that use rc.d based init scripts are currently supported."; \
echo "You may be able to copy one of the scripts in contrib/init.d/ into your /etc/init.d/ directory."; \
exit 1; \
fi
dont-optimize: install
@@ -906,47 +927,3 @@ cleantest:
if cmp -s .cleancount .lastclean ; then echo ; else \
$(MAKE) clean; cp -f .cleancount .lastclean;\
fi
patchlist:
@echo "Experimental Patches:"
@for x in patches/*; do \
patch=`basename $$x`; \
if [ "$$patch" = "CVS" ]; then \
continue; \
fi; \
if grep -q ^$$patch$$ patches/.applied; then \
echo "$$patch (applied)"; \
else \
echo "$$patch (available)"; \
fi; \
done
apply:
@if [ -z "$(PATCH)" ]; then \
echo "Usage: make PATCH=<patchname> apply"; \
elif grep -q ^$(PATCH)$$ patches/.applied 2>/dev/null; then \
echo "Patch $(PATCH) is already applied"; \
elif [ -f "patches/$(PATCH)" ]; then \
echo "Applying patch $(PATCH)"; \
patch -p0 < patches/$(PATCH); \
echo "$(PATCH)" >> patches/.applied; \
else \
echo "No such patch $(PATCH) in patches directory"; \
fi
unapply:
@if [ -z "$(PATCH)" ]; then \
echo "Usage: make PATCH=<patchname> unapply"; \
elif grep -v -q ^$(PATCH)$$ patches/.applied 2>/dev/null; then \
echo "Patch $(PATCH) is not applied"; \
elif [ -f "patches/$(PATCH)" ]; then \
echo "Un-applying patch $(PATCH)"; \
patch -p0 -R < patches/$(PATCH); \
rm -f patches/.tmpapplied || :; \
mv patches/.applied patches/.tmpapplied; \
cat patches/.tmpapplied | grep -v ^$(PATCH)$$ > patches/.applied; \
rm -f patches/.tmpapplied; \
else \
echo "No such patch $(PATCH) in patches directory"; \
fi

10
README Executable file → Normal file
View File

@@ -67,7 +67,7 @@ made every effort possible to maintain backwards compatibility.
In order to discover new features to use, please check the configuration
examples in the /configs directory of the source code distribution.
To discover the major new features of Asterisk 1.2, please visit
http://www.astricon.net/asterisk1-2/
http://edvina.net/asterisk1-2/
* NEW INSTALLATIONS
@@ -164,6 +164,14 @@ parameters. For example, the line "channel => 25-47" creates objects for
the channels 25 through 47 of the card, obtaining the settings
from the variables specified above.
* NETWORK SECURITY DEVICES
This release of Asterisk contains support for network security devices
manufactured by Ranch Networks, Inc., using their MIDCOM interface
library. You will need the companion libmidcom-0.1.0.tar.gz file to
build the library. Contact Ranch Networks' support department for assistance
in building and configuring MIDCOM support.
* SPECIAL NOTE ON TIME
Those using SIP phones should be aware that Asterisk is sensitive to

View File

@@ -1,8 +0,0 @@
About Hold Music
================
Digium has licensed the music included with
the Asterisk distribution From FreePlayMusic
for use and distribution with Asterisk. It
is licensed ONLY for use as hold music within
an Asterisk based PBX.

22
README.opsound Normal file
View File

@@ -0,0 +1,22 @@
About Hold Music
================
These files were obtained from http://opsound.org, where the authors placed them
under the Creative Commons Attribution-Share Alike 2.5 license, a copy of which
may be found at http://creativecommons.org.
Credits
================
macroform-cold_day - Paul Shuler (Macroform)
paulshuler@gmail.com - http://macroform.bandcamp.com/
macroform-robot_dity - Paul Shuler (Macroform)
paulshuler@gmail.com - http://macroform.bandcamp.com/
macroform-the_simplicity - Paul Shuler (Macroform)
paulshuler@gmail.com - http://macroform.bandcamp.com/
manolo_camp-morning_coffee - Manolo Camp
beatbastard@gmx.net - http://ccmixter.org/people/ManoloCamp
reno_project-system - Reno Project
renoproject@hotmail.com - http://www.jamendo.com/en/album/23661

6
SECURITY Executable file → Normal file
View File

@@ -65,3 +65,9 @@ exten => 6123,Dial(Zap/1)
DON'T FORGET TO TAKE THE DEMO CONTEXT OUT OF YOUR DEFAULT CONTEXT. There
isn't really a security reason, it just will keep people from wanting to
play with your Asterisk setup remotely.
* LOG SECURITY
Please note that the Asterisk log files, as well as information printed to the
Asterisk CLI, may contain sensitive information such as passwords and call
history. Keep this in mind when providing access to these resources.

11
UPGRADE.txt Executable file → Normal file
View File

@@ -52,8 +52,10 @@ Dialing:
IAX:
* The naming convention for IAX channels has changed in a minor way such
that the call number follows a "-" rather than a "/" character.
* The naming convention for IAX channels has changed in two ways:
1. The call number follows a "-" rather than a "/" character.
2. The name of the channel has been simplified to IAX2/peer-callno,
rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
SIP:
@@ -129,6 +131,11 @@ Applications:
'mailbox options' menu, and 'change your password' option has been
moved to option 5.
* The application VoiceMailMain now only matches the 'default' context if
none is specified in the arguments. (This was the previously
documented behavior, however, we didn't follow that behavior.) The old
behavior can be restored by setting searchcontexts=yes in voicemail.conf.
Queues:
* A queue is now considered empty not only if there are no members but if

4
acl.c Executable file → Normal file
View File

@@ -176,12 +176,12 @@ struct ast_ha *ast_append_ha(char *sense, char *stuff, struct ast_ha *path)
} else if (!inet_aton(nm, &ha->netmask)) {
ast_log(LOG_WARNING, "%s is not a valid netmask\n", nm);
free(ha);
return path;
return ret;
}
if (!inet_aton(tmp, &ha->netaddr)) {
ast_log(LOG_WARNING, "%s is not a valid IP\n", tmp);
free(ha);
return path;
return ret;
}
ha->netaddr.s_addr &= ha->netmask.s_addr;
if (!strncasecmp(sense, "p", 1)) {

0
aescrypt.c Executable file → Normal file
View File

0
aeskey.c Executable file → Normal file
View File

0
aesopt.h Executable file → Normal file
View File

0
aestab.c Executable file → Normal file
View File

View File

@@ -1,3 +0,0 @@
eagi-test
eagi-sphinx-test
.depend

0
agi/DialAnMp3.agi Executable file → Normal file
View File

2
agi/Makefile Executable file → Normal file
View File

@@ -13,7 +13,7 @@
AGIS=agi-test.agi eagi-test eagi-sphinx-test
CFLAGS+=
CFLAGS+=-DNO_AST_MM
LIBS=
ifeq ($(OSARCH),SunOS)

0
agi/agi-test.agi Executable file → Normal file
View File

0
agi/eagi-sphinx-test.c Executable file → Normal file
View File

0
agi/eagi-test.c Executable file → Normal file
View File

0
agi/fastagi-test Executable file → Normal file
View File

0
agi/numeralize Executable file → Normal file
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0
alaw.c Executable file → Normal file
View File

259
app.c Executable file → Normal file
View File

@@ -48,9 +48,11 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/indications.h"
#include "asterisk/linkedlists.h"
#define MAX_OTHER_FORMATS 10
static AST_LIST_HEAD_STATIC(groups, ast_group_info);
/* !
This function presents a dialtone and reads an extension into 'collect'
@@ -84,14 +86,11 @@ int ast_app_dtget(struct ast_channel *chan, const char *context, char *collect,
ast_playtones_stop(chan);
if (res < 1)
break;
collect[x++] = res;
if (!ast_matchmore_extension(chan, context, collect, 1, chan->cid.cid_num)) {
if (collect[x-1] == '#') {
/* Not a valid extension, ending in #, assume the # was to finish dialing */
collect[x-1] = '\0';
}
if (res == '#')
break;
collect[x++] = res;
if (!ast_matchmore_extension(chan, context, collect, 1, chan->cid.cid_num))
break;
}
}
if (res >= 0) {
if (ast_exists_extension(chan, context, collect, 1, chan->cid.cid_num))
@@ -316,8 +315,12 @@ int ast_dtmf_stream(struct ast_channel *chan,struct ast_channel *peer,char *digi
}
}
}
if (peer)
res = ast_autoservice_stop(peer);
if (peer) {
/* Stop autoservice on the peer channel, but don't overwrite any error condition
that has occurred previously while acting on the primary channel */
if (ast_autoservice_stop(peer) && !res)
res = -1;
}
}
return res;
}
@@ -430,11 +433,11 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
const char *stop, const char *pause,
const char *restart, int skipms)
{
long elapsed = 0, last_elapsed = 0;
char *breaks = NULL;
char *end = NULL;
int blen = 2;
int res;
long pause_restart_point = 0;
if (stop)
blen += strlen(stop);
@@ -456,9 +459,6 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
if (chan->_state != AST_STATE_UP)
res = ast_answer(chan);
if (chan)
ast_stopstream(chan);
if (file) {
if ((end = strchr(file,':'))) {
if (!strcasecmp(end, ":end")) {
@@ -469,25 +469,18 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
}
for (;;) {
struct timeval started = ast_tvnow();
if (chan)
ast_stopstream(chan);
ast_stopstream(chan);
res = ast_streamfile(chan, file, chan->language);
if (!res) {
if (end) {
if (pause_restart_point) {
ast_seekstream(chan->stream, pause_restart_point, SEEK_SET);
pause_restart_point = 0;
}
else if (end) {
ast_seekstream(chan->stream, 0, SEEK_END);
end=NULL;
}
res = 1;
if (elapsed) {
ast_stream_fastforward(chan->stream, elapsed);
last_elapsed = elapsed - 200;
}
if (res)
res = ast_waitstream_fr(chan, breaks, fwd, rev, skipms);
else
break;
end = NULL;
};
res = ast_waitstream_fr(chan, breaks, fwd, rev, skipms);
}
if (res < 1)
@@ -496,17 +489,16 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
/* We go at next loop if we got the restart char */
if (restart && strchr(restart, res)) {
ast_log(LOG_DEBUG, "we'll restart the stream here at next loop\n");
elapsed=0; /* To make sure the next stream will start at beginning */
pause_restart_point = 0;
continue;
}
if (pause != NULL && strchr(pause, res)) {
elapsed = ast_tvdiff_ms(ast_tvnow(), started) + last_elapsed;
for(;;) {
if (chan)
ast_stopstream(chan);
if (pause && strchr(pause, res)) {
pause_restart_point = ast_tellstream(chan->stream);
for (;;) {
ast_stopstream(chan);
res = ast_waitfordigit(chan, 1000);
if (res == 0)
if (!res)
continue;
else if (res == -1 || strchr(pause, res) || (stop && strchr(stop, res)))
break;
@@ -516,17 +508,16 @@ int ast_control_streamfile(struct ast_channel *chan, const char *file,
continue;
}
}
if (res == -1)
break;
/* if we get one of our stop chars, return it to the calling function */
if (stop && strchr(stop, res)) {
/* res = 0; */
if (stop && strchr(stop, res))
break;
}
}
if (chan)
ast_stopstream(chan);
ast_stopstream(chan);
return res;
}
@@ -545,7 +536,7 @@ int ast_play_and_wait(struct ast_channel *chan, const char *fn)
static int global_silence_threshold = 128;
static int global_maxsilence = 0;
int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int silencethreshold, int maxsilence, const char *path)
int ast_play_and_record_full(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int silencethreshold, int maxsilence, const char *path, const char *acceptdtmf, const char *canceldtmf)
{
int d;
char *fmts;
@@ -559,7 +550,6 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
struct ast_dsp *sildet=NULL; /* silence detector dsp */
int totalsilence = 0;
int dspsilence = 0;
int gotsilence = 0; /* did we timeout for silence? */
int rfmt=0;
struct ast_silence_generator *silgen = NULL;
@@ -597,7 +587,7 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
while((fmt = strsep(&stringp, "|"))) {
if (fmtcnt > MAX_OTHER_FORMATS - 1) {
ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app_voicemail.c\n");
ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app.c\n");
break;
}
sfmt[fmtcnt++] = ast_strdupa(fmt);
@@ -682,7 +672,7 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Recording automatically stopped after a silence of %d seconds\n", totalsilence/1000);
ast_frfree(f);
gotsilence = 1;
res = 'S';
outmsg=2;
break;
}
@@ -697,19 +687,18 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
/* Write only once */
ast_writestream(others[0], f);
} else if (f->frametype == AST_FRAME_DTMF) {
if (f->subclass == '#') {
if (strchr(acceptdtmf, f->subclass)) {
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
res = '#';
ast_verbose(VERBOSE_PREFIX_3 "User ended message by pressing %c\n", f->subclass);
res = f->subclass;
outmsg = 2;
ast_frfree(f);
break;
}
if (f->subclass == '0') {
/* Check for a '0' during message recording also, in case caller wants operator */
if (strchr(canceldtmf, f->subclass)) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "User cancelled by pressing %c\n", f->subclass);
res = '0';
ast_verbose(VERBOSE_PREFIX_3 "User cancelled message by pressing %c\n", f->subclass);
res = f->subclass;
outmsg = 0;
ast_frfree(f);
break;
@@ -771,6 +760,14 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
return res;
}
static char default_acceptdtmf[] = "#";
static char default_canceldtmf[] = "0";
int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, int silencethreshold, int maxsilence, const char *path)
{
return ast_play_and_record_full(chan, playfile, recordfile, maxtime, fmt, duration, silencethreshold, maxsilence, path, default_acceptdtmf, default_canceldtmf);
}
int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordfile, int maxtime, char *fmt, int *duration, int beep, int silencethreshold, int maxsilence)
{
int d = 0;
@@ -786,7 +783,6 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
struct ast_dsp *sildet; /* silence detector dsp */
int totalsilence = 0;
int dspsilence = 0;
int gotsilence = 0; /* did we timeout for silence? */
int rfmt=0;
char prependfile[80];
@@ -827,7 +823,7 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
while((fmt = strsep(&stringp, "|"))) {
if (fmtcnt > MAX_OTHER_FORMATS - 1) {
ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app_voicemail.c\n");
ast_log(LOG_WARNING, "Please increase MAX_OTHER_FORMATS in app.c\n");
break;
}
sfmt[fmtcnt++] = ast_strdupa(fmt);
@@ -855,6 +851,7 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
ast_dsp_free(sildet);
return -1;
}
}
@@ -904,7 +901,7 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Recording automatically stopped after a silence of %d seconds\n", totalsilence/1000);
ast_frfree(f);
gotsilence = 1;
res = 'S';
outmsg=2;
break;
}
@@ -959,6 +956,7 @@ int ast_play_and_prepend(struct ast_channel *chan, char *playfile, char *recordf
} else {
ast_log(LOG_WARNING, "Error creating writestream '%s', format '%s'\n", prependfile, sfmt[x]);
}
ast_dsp_free(sildet);
*duration = end - start;
#if 0
if (outmsg > 1) {
@@ -1027,61 +1025,78 @@ int ast_app_group_split_group(char *data, char *group, int group_max, char *cate
else
res = -1;
if (cat)
snprintf(category, category_max, "%s_%s", GROUP_CATEGORY_PREFIX, cat);
else
ast_copy_string(category, GROUP_CATEGORY_PREFIX, category_max);
if (!ast_strlen_zero(cat))
ast_copy_string(category, cat, category_max);
return res;
}
int ast_app_group_set_channel(struct ast_channel *chan, char *data)
{
int res=0;
char group[80] = "";
char category[80] = "";
int res = 0;
char group[80] = "", category[80] = "";
struct ast_group_info *gi = NULL;
size_t len = 0;
if (!ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category))) {
pbx_builtin_setvar_helper(chan, category, group);
} else
if (ast_app_group_split_group(data, group, sizeof(group), category, sizeof(category)))
return -1;
/* Calculate memory we will need if this is new */
len = sizeof(*gi) + strlen(group) + 1;
if (!ast_strlen_zero(category))
len += strlen(category) + 1;
AST_LIST_LOCK(&groups);
AST_LIST_TRAVERSE_SAFE_BEGIN(&groups, gi, list) {
if ((gi->chan == chan) && ((ast_strlen_zero(category) && ast_strlen_zero(gi->category)) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category)))) {
AST_LIST_REMOVE_CURRENT(&groups, list);
free(gi);
break;
}
}
AST_LIST_TRAVERSE_SAFE_END
if ((gi = calloc(1, len))) {
gi->chan = chan;
gi->group = (char *) gi + sizeof(*gi);
strcpy(gi->group, group);
if (!ast_strlen_zero(category)) {
gi->category = (char *) gi + sizeof(*gi) + strlen(group) + 1;
strcpy(gi->category, category);
}
AST_LIST_INSERT_TAIL(&groups, gi, list);
} else {
res = -1;
}
AST_LIST_UNLOCK(&groups);
return res;
}
int ast_app_group_get_count(char *group, char *category)
{
struct ast_channel *chan;
struct ast_group_info *gi = NULL;
int count = 0;
char *test;
char cat[80];
char *s;
if (ast_strlen_zero(group))
return 0;
s = (!ast_strlen_zero(category)) ? category : GROUP_CATEGORY_PREFIX;
ast_copy_string(cat, s, sizeof(cat));
chan = NULL;
while ((chan = ast_channel_walk_locked(chan)) != NULL) {
test = pbx_builtin_getvar_helper(chan, cat);
if (test && !strcasecmp(test, group))
count++;
ast_mutex_unlock(&chan->lock);
AST_LIST_LOCK(&groups);
AST_LIST_TRAVERSE(&groups, gi, list) {
if (!strcasecmp(gi->group, group) && (ast_strlen_zero(category) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category))))
count++;
}
AST_LIST_UNLOCK(&groups);
return count;
}
int ast_app_group_match_get_count(char *groupmatch, char *category)
{
struct ast_group_info *gi = NULL;
regex_t regexbuf;
struct ast_channel *chan;
int count = 0;
char *test;
char cat[80];
char *s;
if (ast_strlen_zero(groupmatch))
return 0;
@@ -1090,22 +1105,64 @@ int ast_app_group_match_get_count(char *groupmatch, char *category)
if (regcomp(&regexbuf, groupmatch, REG_EXTENDED | REG_NOSUB))
return 0;
s = (!ast_strlen_zero(category)) ? category : GROUP_CATEGORY_PREFIX;
ast_copy_string(cat, s, sizeof(cat));
chan = NULL;
while ((chan = ast_channel_walk_locked(chan)) != NULL) {
test = pbx_builtin_getvar_helper(chan, cat);
if (test && !regexec(&regexbuf, test, 0, NULL, 0))
AST_LIST_LOCK(&groups);
AST_LIST_TRAVERSE(&groups, gi, list) {
if (!regexec(&regexbuf, gi->group, 0, NULL, 0) && (ast_strlen_zero(category) || (!ast_strlen_zero(gi->category) && !strcasecmp(gi->category, category))))
count++;
ast_mutex_unlock(&chan->lock);
}
AST_LIST_UNLOCK(&groups);
regfree(&regexbuf);
return count;
}
int ast_app_group_update(struct ast_channel *old, struct ast_channel *new)
{
struct ast_group_info *gi = NULL;
AST_LIST_LOCK(&groups);
AST_LIST_TRAVERSE(&groups, gi, list) {
if (gi->chan == old)
gi->chan = new;
}
AST_LIST_UNLOCK(&groups);
return 0;
}
int ast_app_group_discard(struct ast_channel *chan)
{
struct ast_group_info *gi = NULL;
AST_LIST_LOCK(&groups);
AST_LIST_TRAVERSE_SAFE_BEGIN(&groups, gi, list) {
if (gi->chan == chan) {
AST_LIST_REMOVE_CURRENT(&groups, list);
free(gi);
}
}
AST_LIST_TRAVERSE_SAFE_END
AST_LIST_UNLOCK(&groups);
return 0;
}
int ast_app_group_list_lock(void)
{
return AST_LIST_LOCK(&groups);
}
struct ast_group_info *ast_app_group_list_head(void)
{
return AST_LIST_FIRST(&groups);
}
int ast_app_group_list_unlock(void)
{
return AST_LIST_UNLOCK(&groups);
}
unsigned int ast_app_separate_args(char *buf, char delim, char **array, int arraylen)
{
int argc;
@@ -1159,7 +1216,7 @@ enum AST_LOCK_RESULT ast_lock_path(const char *path)
snprintf(fs, strlen(path) + 19, "%s/.lock-%08x", path, rand());
fd = open(fs, O_WRONLY | O_CREAT | O_EXCL, 0600);
if (fd < 0) {
fprintf(stderr, "Unable to create lock file '%s': %s\n", path, strerror(errno));
ast_log(LOG_ERROR, "Unable to create lock file '%s': %s\n", path, strerror(errno));
return AST_LOCK_PATH_NOT_FOUND;
}
close(fd);
@@ -1168,11 +1225,13 @@ enum AST_LOCK_RESULT ast_lock_path(const char *path)
time(&start);
while (((res = link(fs, s)) < 0) && (errno == EEXIST) && (time(NULL) - start < 5))
usleep(1);
unlink(fs);
if (res) {
ast_log(LOG_WARNING, "Failed to lock path '%s': %s\n", path, strerror(errno));
return AST_LOCK_TIMEOUT;
} else {
unlink(fs);
ast_log(LOG_DEBUG, "Locked path '%s'\n", path);
return AST_LOCK_SUCCESS;
}
@@ -1181,12 +1240,22 @@ enum AST_LOCK_RESULT ast_lock_path(const char *path)
int ast_unlock_path(const char *path)
{
char *s;
int res;
s = alloca(strlen(path) + 10);
if (!s)
if (!s) {
ast_log(LOG_WARNING, "Out of memory!\n");
return -1;
}
snprintf(s, strlen(path) + 9, "%s/%s", path, ".lock");
ast_log(LOG_DEBUG, "Unlocked path '%s'\n", path);
return unlink(s);
if ((res = unlink(s)))
ast_log(LOG_ERROR, "Could not unlock path '%s': %s\n", path, strerror(errno));
else
ast_log(LOG_DEBUG, "Unlocked path '%s'\n", path);
return res;
}
int ast_record_review(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime, const char *fmt, int *duration, const char *path)

View File

@@ -1 +0,0 @@
.depend

7
apps/Makefile Executable file → Normal file
View File

@@ -86,7 +86,7 @@ endif
all: $(APPS)
clean:
rm -f *.so *.o look .depend
rm -f *.so *.o .depend
%.so : %.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB}
@@ -103,7 +103,7 @@ app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} $(CURLLIBS)
app_sql_postgres.o: app_sql_postgres.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres.o app_sql_postgres.c
$(CC) -pipe -I$(CROSS_COMPILE_TARGET)/usr/local/pgsql/include -I$(CROSS_COMPILE_TARGET)/usr/include/postgresql $(CFLAGS) -c -o app_sql_postgres.o app_sql_postgres.c
app_sql_postgres.so: app_sql_postgres.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -L/usr/local/pgsql/lib -lpq
@@ -111,9 +111,6 @@ app_sql_postgres.so: app_sql_postgres.o
app_sql_odbc.so: app_sql_odbc.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -lodbc
look: look.c
$(CC) -pipe -O6 -g look.c -o look -lncurses
ifeq (SunOS,$(shell uname))
app_chanspy.so: app_chanspy.o
$(CC) $(SOLINK) -o $@ $< -lrt

0
apps/app_adsiprog.c Executable file → Normal file
View File

7
apps/app_alarmreceiver.c Executable file → Normal file
View File

@@ -145,7 +145,7 @@ static void database_increment( char *key )
res = ast_db_put(db_family, key, value);
if((res)&&(option_verbose >= 4))
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: database_increment write error");
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: database_increment write error\n");
return;
}
@@ -215,6 +215,7 @@ static int send_tone_burst(struct ast_channel *chan, float freq, int duration, i
i += wf.datalen / 8;
if (i > duration) {
ast_frfree(f);
break;
}
if (ast_write(chan, &wf)){
@@ -222,6 +223,7 @@ static int send_tone_burst(struct ast_channel *chan, float freq, int duration, i
ast_verbose(VERBOSE_PREFIX_4 "AlarmReceiver: Failed to write frame on %s\n", chan->name);
ast_log(LOG_WARNING, "AlarmReceiver Failed to write frame on %s\n",chan->name);
res = -1;
ast_frfree(f);
break;
}
}
@@ -558,11 +560,10 @@ static int receive_ademco_contact_id( struct ast_channel *chan, void *data, int
if(checksum){
database_increment("checksum-errors");
if(option_verbose >= 2){
if(option_verbose >= 2)
ast_verbose(VERBOSE_PREFIX_2 "AlarmReceiver: Nonzero checksum\n");
ast_log(LOG_DEBUG, "AlarmReceiver: Nonzero checksum\n");
continue;
}
}
/* Check the message type for correctness */

2
apps/app_authenticate.c Executable file → Normal file
View File

@@ -186,7 +186,7 @@ static int auth_exec(struct ast_channel *chan, void *data)
if (!res)
res = ast_waitstream(chan, "");
} else {
if (jump && ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
if (jump && ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101) == 0) {
res = 0;
} else {
if (!ast_streamfile(chan, "vm-goodbye", chan->language))

0
apps/app_cdr.c Executable file → Normal file
View File

0
apps/app_chanisavail.c Executable file → Normal file
View File

206
apps/app_chanspy.c Executable file → Normal file
View File

@@ -3,7 +3,8 @@
*
* Copyright (C) 2005 Anthony Minessale II (anthmct@yahoo.com)
*
* Disclaimed to Digium
* A license has been granted to Digium (via disclaimer) for the use of
* this code.
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
@@ -52,7 +53,7 @@ AST_MUTEX_DEFINE_STATIC(modlock);
#define ALL_DONE(u, ret) LOCAL_USER_REMOVE(u); return ret;
#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
static const char *synopsis = "Listen to the audio of an active channel\n";
static const char *synopsis = "Listen to the audio of an active channel";
static const char *app = "ChanSpy";
static const char *desc =
" ChanSpy([chanprefix][|options]): This application is used to listen to the\n"
@@ -130,7 +131,7 @@ static struct ast_channel *local_get_channel_begin_name(char *name)
ast_mutex_lock(&modlock);
chan = local_channel_walk(NULL);
while (chan) {
if (!strncmp(chan->name, name, strlen(name))) {
if (!strncmp(chan->name, name, strlen(name)) && strncmp(chan->name, "Zap/pseudo", 10)) {
ret = chan;
break;
}
@@ -206,21 +207,6 @@ static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, s
return res;
}
static void stop_spying(struct ast_channel *chan, struct ast_channel_spy *spy)
{
/* If our status has changed, then the channel we're spying on is gone....
DON'T TOUCH IT!!! RUN AWAY!!! */
if (spy->status != CHANSPY_RUNNING)
return;
if (!chan)
return;
ast_mutex_lock(&chan->lock);
ast_channel_spy_remove(chan, spy);
ast_mutex_unlock(&chan->lock);
};
/* Map 'volume' levels from -4 through +4 into
decibel (dB) settings for channel drivers
*/
@@ -251,102 +237,112 @@ static void set_volume(struct ast_channel *chan, struct chanspy_translation_help
static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int *volfactor, int fd)
{
struct chanspy_translation_helper csth;
int running, res = 0, x = 0;
char inp[24];
char *name=NULL;
struct ast_frame *f;
int running = 0, res = 0, x = 0;
char inp[24] = "", *name = NULL;
struct ast_frame *f = NULL;
running = (chan && !ast_check_hangup(chan) && spyee && !ast_check_hangup(spyee));
if ((chan && ast_check_hangup(chan)) || (spyee && ast_check_hangup(spyee)))
return 0;
if (running) {
memset(inp, 0, sizeof(inp));
name = ast_strdupa(spyee->name);
if (option_verbose >= 2)
ast_verbose(VERBOSE_PREFIX_2 "Spying on channel %s\n", name);
name = ast_strdupa(spyee->name);
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "Spying on channel %s\n", name);
memset(&csth, 0, sizeof(csth));
ast_set_flag(&csth.spy, CHANSPY_FORMAT_AUDIO);
ast_set_flag(&csth.spy, CHANSPY_TRIGGER_NONE);
ast_set_flag(&csth.spy, CHANSPY_MIXAUDIO);
csth.spy.type = chanspy_spy_type;
csth.spy.status = CHANSPY_RUNNING;
csth.spy.read_queue.format = AST_FORMAT_SLINEAR;
csth.spy.write_queue.format = AST_FORMAT_SLINEAR;
ast_mutex_init(&csth.spy.lock);
csth.volfactor = *volfactor;
set_volume(chan, &csth);
memset(&csth, 0, sizeof(csth));
ast_set_flag(&csth.spy, CHANSPY_FORMAT_AUDIO);
ast_set_flag(&csth.spy, CHANSPY_TRIGGER_NONE);
ast_set_flag(&csth.spy, CHANSPY_MIXAUDIO);
csth.spy.type = chanspy_spy_type;
csth.spy.status = CHANSPY_RUNNING;
csth.spy.read_queue.format = AST_FORMAT_SLINEAR;
csth.spy.write_queue.format = AST_FORMAT_SLINEAR;
ast_mutex_init(&csth.spy.lock);
csth.volfactor = *volfactor;
set_volume(chan, &csth);
if (csth.volfactor) {
ast_set_flag(&csth.spy, CHANSPY_READ_VOLADJUST);
csth.spy.read_vol_adjustment = csth.volfactor;
ast_set_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST);
csth.spy.write_vol_adjustment = csth.volfactor;
csth.fd = fd;
}
csth.fd = fd;
if (start_spying(spyee, chan, &csth.spy))
running = 0;
if (start_spying(spyee, chan, &csth.spy)) {
ast_channel_spy_free(&csth.spy);
return 0;
}
if (running) {
running = 1;
ast_activate_generator(chan, &spygen, &csth);
ast_activate_generator(chan, &spygen, &csth);
while (csth.spy.status == CHANSPY_RUNNING &&
chan && !ast_check_hangup(chan) &&
spyee &&
!ast_check_hangup(spyee) &&
running == 1 &&
(res = ast_waitfor(chan, -1) > -1)) {
if ((f = ast_read(chan))) {
res = 0;
if (f->frametype == AST_FRAME_DTMF) {
res = f->subclass;
}
ast_frfree(f);
if (!res) {
continue;
}
} else {
while (csth.spy.status == CHANSPY_RUNNING &&
(res = ast_waitfor(chan, -1) > -1)) {
/* Read in frame from channel, break out if no frame */
if (!(f = ast_read(chan)))
break;
/* Now if this is DTMF then we have to handle it as such, otherwise just skip it */
res = 0;
if (f->frametype == AST_FRAME_DTMF)
res = f->subclass;
ast_frfree(f);
if (!res)
continue;
if (x == sizeof(inp))
x = 0;
if (res < 0) {
running = -1;
break;
}
/* Process DTMF digits */
if (res == '#') {
if (!ast_strlen_zero(inp)) {
running = x ? atoi(inp) : -1;
break;
}
if (x == sizeof(inp)) {
x = 0;
}
if (res < 0) {
running = -1;
}
if (res == 0) {
continue;
} else if (res == '*') {
running = 0;
} else if (res == '#') {
if (!ast_strlen_zero(inp)) {
running = x ? atoi(inp) : -1;
break;
} else {
(*volfactor)++;
if (*volfactor > 4) {
*volfactor = -4;
}
if (option_verbose > 2) {
ast_verbose(VERBOSE_PREFIX_3 "Setting spy volume on %s to %d\n", chan->name, *volfactor);
}
csth.volfactor = *volfactor;
set_volume(chan, &csth);
} else {
(*volfactor)++;
if (*volfactor > 4)
*volfactor = -1;
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Setting spy volume on %s to %d\n", chan->name, *volfactor);
csth.volfactor = *volfactor;
set_volume(chan, &csth);
if (csth.volfactor) {
ast_set_flag(&csth.spy, CHANSPY_READ_VOLADJUST);
csth.spy.read_vol_adjustment = csth.volfactor;
ast_set_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST);
csth.spy.write_vol_adjustment = csth.volfactor;
} else {
ast_clear_flag(&csth.spy, CHANSPY_READ_VOLADJUST);
ast_clear_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST);
}
} else if (res >= 48 && res <= 57) {
inp[x++] = res;
}
} else if (res == '*') {
break;
} else if (res >= 48 && res <= 57) {
inp[x++] = res;
}
ast_deactivate_generator(chan);
stop_spying(spyee, &csth.spy);
if (option_verbose >= 2) {
ast_verbose(VERBOSE_PREFIX_2 "Done Spying on channel %s\n", name);
}
} else {
running = 0;
}
ast_mutex_destroy(&csth.spy.lock);
ast_deactivate_generator(chan);
csth.spy.status = CHANSPY_DONE;
ast_mutex_lock(&csth.spy.lock);
if (csth.spy.chan) {
ast_mutex_lock(&csth.spy.chan->lock);
ast_channel_spy_remove(csth.spy.chan, &csth.spy);
ast_mutex_unlock(&csth.spy.chan->lock);
}
ast_mutex_unlock(&csth.spy.lock);
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "Done Spying on channel %s\n", name);
ast_channel_spy_free(&csth.spy);
return running;
}
@@ -418,29 +414,31 @@ static int chanspy_exec(struct ast_channel *chan, void *data)
char *opts[OPT_ARG_ARRAY_SIZE];
ast_app_parse_options(chanspy_opts, &flags, opts, options);
if (ast_test_flag(&flags, OPTION_GROUP)) {
mygroup = opts[1];
mygroup = opts[OPT_ARG_GROUP];
}
if (ast_test_flag(&flags, OPTION_RECORD)) {
if (!(recbase = opts[2])) {
if (!(recbase = opts[OPT_ARG_RECORD])) {
recbase = "chanspy";
}
}
silent = ast_test_flag(&flags, OPTION_QUIET);
bronly = ast_test_flag(&flags, OPTION_BRIDGED);
if (ast_test_flag(&flags, OPTION_VOLUME) && opts[1]) {
if (ast_test_flag(&flags, OPTION_VOLUME) && opts[OPT_ARG_VOLUME]) {
int vol;
if ((sscanf(opts[0], "%d", &vol) != 1) || (vol > 4) || (vol < -4))
if ((sscanf(opts[OPT_ARG_VOLUME], "%d", &vol) != 1) || (vol > 4) || (vol < -4))
ast_log(LOG_NOTICE, "Volume factor must be a number between -4 and 4\n");
else
volfactor = vol;
}
}
else
ast_clear_flag(&flags, AST_FLAGS_ALL);
if (recbase) {
char filename[512];
snprintf(filename,sizeof(filename),"%s/%s.%ld.raw",ast_config_AST_MONITOR_DIR, recbase, time(NULL));
if ((fd = open(filename, O_CREAT | O_WRONLY, O_TRUNC)) <= 0) {
snprintf(filename,sizeof(filename),"%s/%s.%d.raw",ast_config_AST_MONITOR_DIR, recbase, (int)time(NULL));
if ((fd = open(filename, O_CREAT | O_WRONLY, O_TRUNC, 0644)) <= 0) {
ast_log(LOG_WARNING, "Cannot open %s for recording\n", filename);
fd = 0;
}
@@ -482,7 +480,7 @@ static int chanspy_exec(struct ast_channel *chan, void *data)
}
}
if (igrp && (!spec || ((strlen(spec) < strlen(peer->name) &&
if (igrp && (!spec || ((strlen(spec) <= strlen(peer->name) &&
!strncasecmp(peer->name, spec, strlen(spec)))))) {
if (peer && (!bronly || ast_bridged_channel(peer)) &&
!ast_check_hangup(peer) && !ast_test_flag(peer, AST_FLAG_SPYING)) {

2
apps/app_controlplayback.c Executable file → Normal file
View File

@@ -140,7 +140,7 @@ static int controlplayback_exec(struct ast_channel *chan, void *data)
} else {
if (res < 0) {
if (priority_jump || option_priority_jumping) {
if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101)) {
ast_log(LOG_WARNING, "ControlPlayback tried to jump to priority n+101 as requested, but priority didn't exist\n");
}
}

3
apps/app_curl.c Executable file → Normal file
View File

@@ -94,7 +94,6 @@ static int curl_internal(struct MemoryStruct *chunk, char *url, char *post)
{
CURL *curl;
curl_global_init(CURL_GLOBAL_ALL);
curl = curl_easy_init();
if (!curl) {
@@ -224,6 +223,7 @@ int unload_module(void)
res |= ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
curl_global_cleanup();
return res;
}
@@ -232,6 +232,7 @@ int load_module(void)
{
int res;
curl_global_init(CURL_GLOBAL_ALL);
res = ast_custom_function_register(&acf_curl);
res |= ast_register_application(app, curl_exec, synopsis, descrip);

12
apps/app_cut.c Executable file → Normal file
View File

@@ -36,7 +36,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/version.h"
#include "asterisk/app.h"
/* Maximum length of any variable */
@@ -49,7 +48,7 @@ static char *app_cut = "Cut";
static char *cut_synopsis = "Splits a variable's contents using the specified delimiter";
static char *cut_descrip =
" Cut(newvar=varname,delimiter,fieldspec): This applicaiton will split the\n"
" Cut(newvar=varname,delimiter,fieldspec): This application will split the\n"
"contents of a variable based on the given delimeter and store the result in\n"
"a new variable.\n"
"Parameters:\n"
@@ -67,7 +66,7 @@ static char *app_sort_descrip =
" Sort(newvar=key1:val1[,key2:val2[[...],keyN:valN]]): This application will\n"
"sort the list provided in ascending order. The result will be stored in the\n"
"specified variable name.\n"
" This applicaiton has been deprecated in favor of the SORT function.\n";
" This application has been deprecated in favor of the SORT function.\n";
STANDARD_LOCAL_USER;
@@ -146,8 +145,9 @@ static int sort_internal(struct ast_channel *chan, char *data, char *buffer, siz
int blen = strlen(buffer);
if (element_count++) {
strncat(buffer + blen, ",", buflen - blen - 1);
blen++;
}
strncat(buffer + blen + 1, sortable_keys[count2].key, buflen - blen - 2);
strncat(buffer + blen, sortable_keys[count2].key, buflen - blen - 1);
}
return 0;
@@ -253,6 +253,8 @@ static int cut_internal(struct ast_channel *chan, char *data, char *buffer, size
}
}
}
} else {
return ERROR_NOARG;
}
return 0;
}
@@ -386,7 +388,7 @@ static char *acf_cut_exec(struct ast_channel *chan, char *cmd, char *data, char
switch (cut_internal(chan, data, buf, len)) {
case ERROR_NOARG:
ast_log(LOG_ERROR, "Cut() requires an argument\n");
ast_log(LOG_ERROR, "CUT() requires an argument\n");
break;
case ERROR_NOMEM:
ast_log(LOG_ERROR, "Out of memory\n");

0
apps/app_datetime.c Executable file → Normal file
View File

2
apps/app_db.c Executable file → Normal file
View File

@@ -63,7 +63,7 @@ static char *p_descrip =
" This application has been deprecated in favor of the DB function.\n";
static char *d_descrip =
" DBdel(family/key): This applicaiton will delete a key from the Asterisk\n"
" DBdel(family/key): This application will delete a key from the Asterisk\n"
"database.\n";
static char *dt_descrip =

179
apps/app_dial.c Executable file → Normal file
View File

@@ -1,7 +1,7 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
* Copyright (C) 1999 - 2006, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
@@ -31,6 +31,7 @@
#include <stdio.h>
#include <sys/time.h>
#include <sys/signal.h>
#include <sys/stat.h>
#include <netinet/in.h>
#include "asterisk.h"
@@ -64,7 +65,7 @@ static char *synopsis = "Place a call and connect to the current channel";
static char *descrip =
" Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]):\n"
"This applicaiton will place calls to one or more specified channels. As soon\n"
"This application will place calls to one or more specified channels. As soon\n"
"as one of the requested channels answers, the originating channel will be\n"
"answered, if it has not already been answered. These two channels will then\n"
"be active in a bridged call. All other channels that were requested will then\n"
@@ -75,11 +76,11 @@ static char *descrip =
"continue if no requested channels can be called, or if the timeout expires.\n\n"
" This application sets the following channel variables upon completion:\n"
" DIALEDTIME - This is the time from dialing a channel until when it\n"
" answers.\n"
" is disconnected.\n"
" ANSWEREDTIME - This is the amount of time for actual call.\n"
" DIALSTATUS - This is the status of the call:\n"
" CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n"
" DONTCALL | TORTURE\n"
" DONTCALL | TORTURE | INVALIDARGS\n"
" For the Privacy and Screening Modes, the DIALSTATUS variable will be set to\n"
"DONTCALL if the called party chooses to send the calling party to the 'Go Away'\n"
"script. The DIALSTATUS variable will be set to TORTURE if the called party\n"
@@ -89,7 +90,10 @@ static char *descrip =
"ends the call.\n"
" The optional URL will be sent to the called party if the channel supports it.\n"
" If the OUTBOUND_GROUP variable is set, all peer channels created by this\n"
"application will be put into that group (as in Set(GROUP()=...).\n\n"
"application will be put into that group (as in Set(GROUP()=...).\n"
" If the OUTBOUND_GROUP_ONCE variable is set, all peer channels created by this\n"
"application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,\n"
"however, the variable will be unset after use.\n\n"
" Options:\n"
" A(x) - Play an announcement to the called party, using 'x' as the file.\n"
" C - Reset the CDR for this call.\n"
@@ -108,9 +112,11 @@ static char *descrip =
" other than the number assigned to the caller.\n"
" g - Proceed with dialplan execution at the current extension if the\n"
" destination channel hangs up.\n"
" G(context^exten^pri) - If the call is answered, transfer both parties to\n"
" the specified priority. Optionally, an extension, or extension and\n"
" context may be specified. Otherwise, the current extension is used.\n"
" G(context^exten^pri) - If the call is answered, transfer the calling party to\n"
" the specified priority and the called party to the specified priority+1.\n"
" Optionally, an extension, or extension and context may be specified. \n"
" Otherwise, the current extension is used. You cannot use any additional\n"
" action post answer options in conjunction with this option.\n"
" h - Allow the called party to hang up by sending the '*' DTMF digit.\n"
" H - Allow the calling party to hang up by hitting the '*' DTMF digit.\n"
" j - Jump to priority n+101 if all of the requested channels were busy.\n"
@@ -143,6 +149,8 @@ static char *descrip =
" * GOTO:<context>^<exten>^<priority> - Transfer the call to the\n"
" specified priority. Optionally, an extension, or\n"
" extension and priority can be specified.\n"
" You cannot use any additional action post answer options in conjunction\n"
" with this option.\n"
" n - This option is a modifier for the screen/privacy mode. It specifies\n"
" that no introductions are to be saved in the priv-callerintros\n"
" directory.\n"
@@ -161,7 +169,7 @@ static char *descrip =
" S(x) - Hang up the call after 'x' seconds *after* the called party has\n"
" answered the call.\n"
" t - Allow the called party to transfer the calling party by sending the\n"
" DTMF sequence defiend in features.conf.\n"
" DTMF sequence defined in features.conf.\n"
" T - Allow the calling party to transfer the called party by sending the\n"
" DTMF sequence defined in features.conf.\n"
" w - Allow the called party to enable recording of the call by sending\n"
@@ -303,6 +311,8 @@ static void hanguptree(struct localuser *outgoing, struct ast_channel *exception
ast_cdr_failed(chan->cdr); \
numnochan++; \
break; \
case AST_CAUSE_NORMAL_CLEARING: \
break; \
default: \
numnochan++; \
break; \
@@ -463,7 +473,11 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
ast_verbose(VERBOSE_PREFIX_3 "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, o->chan->name);
/* Setup parameters */
o->chan = ast_request(tech, in->nativeformats, stuff, &cause);
if (!o->chan)
if (o->chan) {
if (single)
ast_channel_make_compatible(o->chan, in);
ast_channel_inherit_variables(in, o->chan);
} else
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
} else {
if (option_verbose > 2)
@@ -512,10 +526,8 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
if (in->cid.cid_ani) {
if (o->chan->cid.cid_ani)
free(o->chan->cid.cid_ani);
o->chan->cid.cid_ani = malloc(strlen(in->cid.cid_ani) + 1);
if (o->chan->cid.cid_ani)
ast_copy_string(o->chan->cid.cid_ani, in->cid.cid_ani, sizeof(o->chan->cid.cid_ani));
else
o->chan->cid.cid_ani = strdup(in->cid.cid_ani);
if (!o->chan->cid.cid_ani)
ast_log(LOG_WARNING, "Out of memory\n");
}
if (o->chan->cid.cid_rdnis)
@@ -650,6 +662,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
ast_hangup(o->chan);
o->chan = NULL;
ast_clear_flag(o, DIAL_STILLGOING);
HANDLE_CAUSE(in->hangupcause, in);
}
}
o = o->next;
@@ -665,6 +678,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
/* Got hung up */
*to=-1;
ast_cdr_noanswer(in->cdr);
strcpy(status, "CANCEL");
if (f)
ast_frfree(f);
@@ -678,6 +692,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
if (option_verbose > 3)
ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
*to=0;
ast_cdr_noanswer(in->cdr);
*result = f->subclass;
strcpy(status, "CANCEL");
ast_frfree(f);
@@ -690,6 +705,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
if (option_verbose > 3)
ast_verbose(VERBOSE_PREFIX_3 "User hit %c to disconnect call.\n", f->subclass);
*to=0;
ast_cdr_noanswer(in->cdr);
strcpy(status, "CANCEL");
ast_frfree(f);
return NULL;
@@ -714,13 +730,17 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
}
if (!*to && (option_verbose > 2))
ast_verbose( VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", orig);
if (!*to || ast_check_hangup(in)) {
ast_cdr_noanswer(in->cdr);
}
}
return peer;
}
static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags *peerflags)
static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags *peerflags, int *continue_exec)
{
int res=-1;
struct localuser *u;
@@ -737,6 +757,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
char numsubst[AST_MAX_EXTENSION];
char restofit[AST_MAX_EXTENSION];
char cidname[AST_MAX_EXTENSION];
char toast[80];
char *newnum;
char *l;
int privdb_val=0;
@@ -750,7 +771,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
char *start_sound=NULL;
char *dtmfcalled=NULL, *dtmfcalling=NULL;
char *var;
char status[256];
char status[256] = "INVALIDARGS";
int play_to_caller=0,play_to_callee=0;
int sentringing=0, moh=0;
char *outbound_group = NULL;
@@ -771,21 +792,19 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
return -1;
}
LOCAL_USER_ADD(u);
if (!(parse = ast_strdupa(data))) {
ast_log(LOG_WARNING, "Memory allocation failure\n");
LOCAL_USER_REMOVE(u);
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (!ast_strlen_zero(args.options)) {
if (ast_app_parse_options(dial_exec_options, &opts, opt_args, args.options)) {
pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
LOCAL_USER_REMOVE(u);
return -1;
}
@@ -793,6 +812,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
if (ast_strlen_zero(args.peers)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
LOCAL_USER_REMOVE(u);
return -1;
}
@@ -826,6 +846,21 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
if (!timelimit) {
timelimit = play_to_caller = play_to_callee = play_warning = warning_freq = 0;
warning_sound = NULL;
} else if (play_warning > timelimit) {
/* If the first warning is requested _after_ the entire call would end,
and no warning frequency is requested, then turn off the warning. If
a warning frequency is requested, reduce the 'first warning' time by
that frequency until it falls within the call's total time limit.
*/
if (!warning_freq) {
play_warning = 0;
} else {
while (play_warning > timelimit)
play_warning -= warning_freq;
if (play_warning < 1)
play_warning = warning_freq = 0;
}
}
var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLER");
@@ -916,27 +951,39 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
}
if( privdb_val == AST_PRIVACY_DENY ) {
strcpy(status, "NOANSWER");
ast_verbose( VERBOSE_PREFIX_3 "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
res=0;
goto out;
}
else if( privdb_val == AST_PRIVACY_KILL ) {
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 201);
strcpy(status, "DONTCALL");
if (option_priority_jumping || ast_test_flag(&opts, OPT_PRIORITY_JUMP)) {
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 201);
}
res = 0;
goto out; /* Is this right? */
}
else if( privdb_val == AST_PRIVACY_TORTURE ) {
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 301);
strcpy(status, "TORTURE");
if (option_priority_jumping || ast_test_flag(&opts, OPT_PRIORITY_JUMP)) {
ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 301);
}
res = 0;
goto out; /* is this right??? */
}
else if( privdb_val == AST_PRIVACY_UNKNOWN ) {
/* Get the user's intro, store it in priv-callerintros/$CID,
unless it is already there-- this should be done before the
call is actually dialed */
/* make sure the priv-callerintros dir exists? */
/* make sure the priv-callerintros dir actually exists */
snprintf(privintro, sizeof(privintro), "%s/sounds/priv-callerintros", ast_config_AST_VAR_DIR);
if (mkdir(privintro, 0755) && errno != EEXIST) {
ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(errno));
res = -1;
goto out;
}
snprintf(privintro,sizeof(privintro),"priv-callerintros/%s", privcid);
if( ast_fileexists(privintro,NULL,NULL ) > 0 && strncmp(privcid,"NOCALLERID",10) != 0) {
@@ -953,17 +1000,35 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
"At the tone, please say your name:"
*/
ast_play_and_record(chan, "priv-recordintro", privintro, 4, "gsm", &duration, 128, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
/* don't think we'll need a lock removed, we took care of
conflicts by naming the privintro file */
ast_answer(chan);
res = ast_play_and_record(chan, "priv-recordintro", privintro, 4, "gsm", &duration, 128, 2000, 0); /* NOTE: I've reduced the total time to */
/* 4 sec don't think we'll need a lock removed, we
took care of conflicts by naming the privintro file */
if (res == -1) {
/* Delete the file regardless since they hung up during recording */
ast_filedelete(privintro, NULL);
if( ast_fileexists(privintro,NULL,NULL ) > 0 )
ast_log(LOG_NOTICE,"privacy: ast_filedelete didn't do its job on %s\n", privintro);
else if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Successfully deleted %s intro file\n", privintro);
goto out;
}
}
}
}
if (continue_exec)
*continue_exec = 0;
/* If a channel group has been specified, get it for use when we create peer channels */
outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP");
ast_copy_flags(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID);
if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
outbound_group = ast_strdupa(outbound_group);
pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
} else {
outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP");
}
ast_copy_flags(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP);
cur = args.peers;
do {
/* Remember where to start next time */
@@ -1013,6 +1078,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
cur = rest;
if (!cur)
chan->hangupcause = cause;
free(tmp);
continue;
}
pbx_builtin_setvar_helper(tmp->chan, "DIALEDPEERNUMBER", numsubst);
@@ -1039,6 +1105,8 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
tmp->chan = ast_request(tech, chan->nativeformats, stuff, &cause);
if (!tmp->chan)
ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
else
ast_channel_inherit_variables(chan, tmp->chan);
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Too many forwards from %s\n", tmp->chan->name);
@@ -1049,6 +1117,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
if (!tmp->chan) {
HANDLE_CAUSE(cause, chan);
cur = rest;
free(tmp);
continue;
}
}
@@ -1111,11 +1180,12 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
/* Again, keep going even if there's an error */
if (option_debug)
ast_log(LOG_DEBUG, "ast call on peer returned %d\n", res);
else if (option_verbose > 2)
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Couldn't call %s\n", numsubst);
ast_hangup(tmp->chan);
tmp->chan = NULL;
cur = rest;
free(tmp);
continue;
} else {
senddialevent(chan, tmp->chan);
@@ -1150,6 +1220,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
strcpy(status, "NOANSWER");
if (ast_test_flag(outgoing, OPT_MUSICBACK)) {
moh=1;
ast_indicate(chan, AST_CONTROL_PROGRESS);
ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK]);
} else if (ast_test_flag(outgoing, OPT_RINGBACK)) {
ast_indicate(chan, AST_CONTROL_RINGING);
@@ -1227,18 +1298,18 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
do {
if (!res2)
res2 = ast_play_and_wait(peer,"priv-callpending");
if( res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2>'5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4') ) /* uh, interrupting with a bad answer is ... ignorable! */
if ( res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2>'5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4') ) /* uh, interrupting with a bad answer is ... ignorable! */
res2 = 0;
/* priv-callpending script:
"I have a caller waiting, who introduces themselves as:"
*/
if (!res2)
res2 = ast_play_and_wait(peer,privintro);
if( res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2>'5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4') ) /* uh, interrupting with a bad answer is ... ignorable! */
res2 = ast_play_and_wait(peer, privintro);
if ( res2 < '1' || (ast_test_flag(&opts, OPT_PRIVACY) && res2>'5') || (ast_test_flag(&opts, OPT_SCREENING) && res2 > '4') ) /* uh, interrupting with a bad answer is ... ignorable! */
res2 = 0;
/* now get input from the called party, as to their choice */
if( !res2 ) {
if (!res2) {
if( ast_test_flag(&opts, OPT_PRIVACY) )
res2 = ast_play_and_wait(peer,"priv-callee-options");
if( ast_test_flag(&opts, OPT_SCREENING) )
@@ -1285,6 +1356,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
opt_args[OPT_ARG_PRIVACY], privcid);
ast_privacy_set(opt_args[OPT_ARG_PRIVACY], privcid, AST_PRIVACY_DENY);
}
strcpy(status,"NOANSWER");
if (ast_test_flag(&opts, OPT_MUSICBACK)) {
ast_moh_stop(chan);
} else if (ast_test_flag(&opts, OPT_RINGBACK)) {
@@ -1418,6 +1490,8 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
ast_pbx_start(peer);
hanguptree(outgoing, NULL);
LOCAL_USER_REMOVE(u);
if (continue_exec)
*continue_exec = 1;
return 0;
}
@@ -1500,7 +1574,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
time_t now;
time(&now);
chan->whentohangup = now + calldurationlimit;
peer->whentohangup = now + calldurationlimit;
}
if (!ast_strlen_zero(dtmfcalled)) {
if (option_verbose > 2)
@@ -1515,8 +1589,6 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
}
if (!res) {
char toast[80];
memset(&config,0,sizeof(struct ast_bridge_config));
if (play_to_caller)
ast_set_flag(&(config.features_caller), AST_FEATURE_PLAY_WARNING);
@@ -1560,13 +1632,15 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
}
res = ast_bridge_call(chan,peer,&config);
time(&end_time);
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
pbx_builtin_setvar_helper(chan, "DIALEDTIME", toast);
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - answer_time));
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", toast);
} else
} else {
time(&end_time);
res = -1;
}
snprintf(toast, sizeof(toast), "%ld", (long)(end_time - start_time));
pbx_builtin_setvar_helper(chan, "DIALEDTIME", toast);
if (res != AST_PBX_NO_HANGUP_PEER) {
if (!chan->_softhangup)
@@ -1586,8 +1660,11 @@ out:
pbx_builtin_setvar_helper(chan, "DIALSTATUS", status);
ast_log(LOG_DEBUG, "Exiting with DIALSTATUS=%s.\n", status);
if ((ast_test_flag(peerflags, OPT_GO_ON)) && (!chan->_softhangup) && (res != AST_PBX_KEEPALIVE))
if ((ast_test_flag(peerflags, OPT_GO_ON)) && (!chan->_softhangup) && (res != AST_PBX_KEEPALIVE)) {
if (calldurationlimit)
chan->whentohangup = 0;
res=0;
}
LOCAL_USER_REMOVE(u);
@@ -1597,8 +1674,10 @@ out:
static int dial_exec(struct ast_channel *chan, void *data)
{
struct ast_flags peerflags;
memset(&peerflags, 0, sizeof(peerflags));
return dial_exec_full(chan, data, &peerflags);
return dial_exec_full(chan, data, &peerflags, NULL);
}
static int retrydial_exec(struct ast_channel *chan, void *data)
@@ -1627,7 +1706,7 @@ static int retrydial_exec(struct ast_channel *chan, void *data)
if ((dialdata = strchr(announce, '|'))) {
*dialdata = '\0';
dialdata++;
if ((sleep = atoi(dialdata))) {
if (sscanf(dialdata, "%d", &sleep) == 1) {
sleep *= 1000;
} else {
ast_log(LOG_ERROR, "%s requires the numerical argument <sleep>\n",rapp);
@@ -1637,7 +1716,7 @@ static int retrydial_exec(struct ast_channel *chan, void *data)
if ((dialdata = strchr(dialdata, '|'))) {
*dialdata = '\0';
dialdata++;
if (!(loops = atoi(dialdata))) {
if (sscanf(dialdata, "%d", &loops) != 1) {
ast_log(LOG_ERROR, "%s requires the numerical argument <loops>\n",rapp);
LOCAL_USER_REMOVE(u);
return -1;
@@ -1663,11 +1742,16 @@ static int retrydial_exec(struct ast_channel *chan, void *data)
context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
while (loops) {
int continue_exec;
chan->data = "Retrying";
if (ast_test_flag(chan, AST_FLAG_MOH))
ast_moh_stop(chan);
if ((res = dial_exec_full(chan, dialdata, &peerflags)) == 0) {
res = dial_exec_full(chan, dialdata, &peerflags, &continue_exec);
if (continue_exec)
break;
if (res == 0) {
if (ast_test_flag(&peerflags, OPT_DTMF_EXIT)) {
if (!(res = ast_streamfile(chan, announce, chan->language)))
res = ast_waitstream(chan, AST_DIGIT_ANY);
@@ -1704,7 +1788,6 @@ static int retrydial_exec(struct ast_channel *chan, void *data)
LOCAL_USER_REMOVE(u);
return loops ? res : 0;
}
int unload_module(void)

5
apps/app_dictate.c Executable file → Normal file
View File

@@ -257,7 +257,8 @@ static int dictate_exec(struct ast_channel *chan, void *data)
if (lastop != DFLAG_PLAY) {
lastop = DFLAG_PLAY;
ast_closestream(fs);
fs = ast_openstream(chan, path, chan->language);
if (!(fs = ast_openstream(chan, path, chan->language)))
break;
ast_seekstream(fs, samples, SEEK_SET);
chan->stream = NULL;
}
@@ -320,7 +321,7 @@ static int dictate_exec(struct ast_channel *chan, void *data)
ast_set_read_format(chan, oldr);
}
LOCAL_USER_REMOVE(u);
return res;
return 0;
}
int unload_module(void)

6
apps/app_directed_pickup.c Executable file → Normal file
View File

@@ -3,7 +3,7 @@
*
* Copyright (C) 2005, Joshua Colp
*
* Joshua Colp <jcolp@asterlink.com>
* Joshua Colp <jcolp@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
@@ -77,7 +77,7 @@ static int pickup_exec(struct ast_channel *chan, void *data)
/* Find a channel to pickup */
origin = ast_get_channel_by_exten_locked(exten, context);
if (origin) {
if (origin && origin->cdr) {
ast_cdr_getvar(origin->cdr, "dstchannel", &tmp, workspace,
sizeof(workspace), 0);
if (tmp) {
@@ -89,6 +89,8 @@ static int pickup_exec(struct ast_channel *chan, void *data)
}
ast_mutex_unlock(&origin->lock);
} else {
if (origin)
ast_mutex_unlock(&origin->lock);
ast_log(LOG_DEBUG, "No originating channel found.\n");
}

170
apps/app_directory.c Executable file → Normal file
View File

@@ -42,6 +42,16 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/say.h"
#include "asterisk/utils.h"
#ifdef USE_ODBC_STORAGE
#include <errno.h>
#include <sys/mman.h>
#include "asterisk/res_odbc.h"
static char odbc_database[80] = "asterisk";
static char odbc_table[80] = "voicemessages";
static char vmfmts[80] = "wav";
#endif
static char *tdesc = "Extension Directory";
static char *app = "Directory";
@@ -51,7 +61,7 @@ static char *descrip =
"the calling channel with a directory of extensions from which they can search\n"
"by name. The list of names and corresponding extensions is retrieved from the\n"
"voicemail configuration file, voicemail.conf.\n"
" This applicaiton will immediate exit if one of the following DTMF digits are\n"
" This application will immediately exit if one of the following DTMF digits are\n"
"received and the extension to jump to exists:\n"
" 0 - Jump to the 'o' extension, if it exists.\n"
" * - Jump to the 'a' extension, if it exists.\n\n"
@@ -77,6 +87,101 @@ STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
#ifdef USE_ODBC_STORAGE
static void retrieve_file(char *dir)
{
int x = 0;
int res;
int fd=-1;
size_t fdlen = 0;
void *fdm = MAP_FAILED;
SQLHSTMT stmt;
char sql[256];
char fmt[80]="", empty[10] = "";
char *c;
SQLLEN colsize;
char full_fn[256];
odbc_obj *obj;
obj = fetch_odbc_obj(odbc_database, 0);
if (obj) {
do {
ast_copy_string(fmt, vmfmts, sizeof(fmt));
c = strchr(fmt, '|');
if (c)
*c = '\0';
if (!strcasecmp(fmt, "wav49"))
strcpy(fmt, "WAV");
snprintf(full_fn, sizeof(full_fn), "%s.%s", dir, fmt);
res = SQLAllocHandle(SQL_HANDLE_STMT, obj->con, &stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Alloc Handle failed!\n");
break;
}
snprintf(sql, sizeof(sql), "SELECT recording FROM %s WHERE dir=? AND msgnum=-1", odbc_table);
res = SQLPrepare(stmt, (unsigned char *)sql, SQL_NTS);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Prepare failed![%s]\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
SQLBindParameter(stmt, 1, SQL_PARAM_INPUT, SQL_C_CHAR, SQL_CHAR, strlen(dir), 0, (void *)dir, 0, NULL);
res = odbc_smart_execute(obj, stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Execute error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
res = SQLFetch(stmt);
if (res == SQL_NO_DATA) {
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
} else if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Fetch error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
fd = open(full_fn, O_RDWR | O_CREAT | O_TRUNC, 0770);
if (fd < 0) {
ast_log(LOG_WARNING, "Failed to write '%s': %s\n", full_fn, strerror(errno));
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
res = SQLGetData(stmt, 1, SQL_BINARY, empty, 0, &colsize);
fdlen = colsize;
if (fd > -1) {
char tmp[1]="";
lseek(fd, fdlen - 1, SEEK_SET);
if (write(fd, tmp, 1) != 1) {
close(fd);
fd = -1;
break;
}
if (fd > -1)
fdm = mmap(NULL, fdlen, PROT_READ | PROT_WRITE, MAP_SHARED, fd, 0);
}
if (fdm != MAP_FAILED) {
memset(fdm, 0, fdlen);
res = SQLGetData(stmt, x + 1, SQL_BINARY, fdm, fdlen, &colsize);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Get Data error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
break;
}
}
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
} while (0);
} else
ast_log(LOG_WARNING, "Failed to obtain database object for '%s'!\n", odbc_database);
if (fdm != MAP_FAILED)
munmap(fdm, fdlen);
if (fd > -1)
close(fd);
return;
}
#endif
static char *convert(char *lastname)
{
char *tmp;
@@ -151,7 +256,7 @@ static char *convert(char *lastname)
* '1' for selected entry from directory
* '*' for skipped entry from directory
*/
static int play_mailbox_owner(struct ast_channel *chan, char *context, char *dialcontext, char *ext, char *name) {
static int play_mailbox_owner(struct ast_channel *chan, char *context, char *dialcontext, char *ext, char *name, int fromappvm) {
int res = 0;
int loop = 3;
char fn[256];
@@ -160,10 +265,16 @@ static int play_mailbox_owner(struct ast_channel *chan, char *context, char *dia
/* Check for the VoiceMail2 greeting first */
snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/greet",
(char *)ast_config_AST_SPOOL_DIR, context, ext);
#ifdef USE_ODBC_STORAGE
retrieve_file(fn);
#endif
/* Otherwise, check for an old-style Voicemail greeting */
snprintf(fn2, sizeof(fn2), "%s/vm/%s/greet",
(char *)ast_config_AST_SPOOL_DIR, ext);
#ifdef USE_ODBC_STORAGE
retrieve_file(fn2);
#endif
if (ast_fileexists(fn, NULL, chan->language) > 0) {
res = ast_streamfile(chan, fn, chan->language);
@@ -181,6 +292,10 @@ static int play_mailbox_owner(struct ast_channel *chan, char *context, char *dia
res = ast_say_character_str(chan, !ast_strlen_zero(name) ? name : ext,
AST_DIGIT_ANY, chan->language);
}
#ifdef USE_ODBC_STORAGE
ast_filedelete(fn, NULL);
ast_filedelete(fn2, NULL);
#endif
while (loop) {
if (!res) {
@@ -199,12 +314,17 @@ static int play_mailbox_owner(struct ast_channel *chan, char *context, char *dia
case '1':
/* Name selected */
loop = 0;
if (ast_goto_if_exists(chan, dialcontext, ext, 1)) {
ast_log(LOG_WARNING,
"Can't find extension '%s' in context '%s'. "
"Did you pass the wrong context to Directory?\n",
ext, dialcontext);
res = -1;
if (fromappvm) {
/* We still want to set the exten */
ast_copy_string(chan->exten, ext, sizeof(chan->exten));
} else {
if (ast_goto_if_exists(chan, dialcontext, ext, 1)) {
ast_log(LOG_WARNING,
"Can't find extension '%s' in context '%s'. "
"Did you pass the wrong context to Directory?\n",
ext, dialcontext);
res = -1;
}
}
break;
@@ -288,7 +408,7 @@ static struct ast_config *realtime_directory(char *context)
return cfg;
}
static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *context, char *dialcontext, char digit, int last)
static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *context, char *dialcontext, char digit, int last, int fromappvm)
{
/* Read in the first three digits.. "digit" is the first digit, already read */
char ext[NUMDIGITS + 1];
@@ -306,7 +426,7 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
return -1;
}
if (digit == '0') {
if (!ast_goto_if_exists(chan, chan->context, "o", 1) ||
if (!ast_goto_if_exists(chan, dialcontext, "o", 1) ||
(!ast_strlen_zero(chan->macrocontext) &&
!ast_goto_if_exists(chan, chan->macrocontext, "o", 1))) {
return 0;
@@ -317,7 +437,7 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
}
}
if (digit == '*') {
if (!ast_goto_if_exists(chan, chan->context, "a", 1) ||
if (!ast_goto_if_exists(chan, dialcontext, "a", 1) ||
(!ast_strlen_zero(chan->macrocontext) &&
!ast_goto_if_exists(chan, chan->macrocontext, "a", 1))) {
return 0;
@@ -350,7 +470,7 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
pos = strrchr(pos, ' ') + 1;
conv = convert(pos);
if (conv) {
if (!strcmp(conv, ext)) {
if (!strncmp(conv, ext, strlen(ext))) {
/* Match! */
found++;
free(conv);
@@ -367,7 +487,7 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
if (v) {
/* We have a match -- play a greeting if they have it */
res = play_mailbox_owner(chan, context, dialcontext, v->name, name);
res = play_mailbox_owner(chan, context, dialcontext, v->name, name, fromappvm);
switch (res) {
case -1:
/* user pressed '1' but extension does not exist, or
@@ -414,6 +534,7 @@ static int directory_exec(struct ast_channel *chan, void *data)
struct localuser *u;
struct ast_config *cfg;
int last = 1;
int fromappvm = 0;
char *context, *dialcontext, *dirintro, *options;
if (ast_strlen_zero(data)) {
@@ -434,6 +555,8 @@ static int directory_exec(struct ast_channel *chan, void *data)
options++;
if (strchr(options, 'f'))
last = 0;
if (strchr(options, 'v'))
fromappvm = 1;
}
} else
dialcontext = context;
@@ -466,7 +589,7 @@ static int directory_exec(struct ast_channel *chan, void *data)
if (!res)
res = ast_waitfordigit(chan, 5000);
if (res > 0) {
res = do_directory(chan, cfg, context, dialcontext, res, last);
res = do_directory(chan, cfg, context, dialcontext, res, last, fromappvm);
if (res > 0) {
res = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
@@ -495,6 +618,25 @@ int unload_module(void)
int load_module(void)
{
#ifdef USE_ODBC_STORAGE
struct ast_config *cfg = ast_config_load(VOICEMAIL_CONFIG);
char *tmp;
if (cfg) {
if ((tmp = ast_variable_retrieve(cfg, "general", "odbcstorage"))) {
ast_copy_string(odbc_database, tmp, sizeof(odbc_database));
}
if ((tmp = ast_variable_retrieve(cfg, "general", "odbctable"))) {
ast_copy_string(odbc_table, tmp, sizeof(odbc_table));
}
if ((tmp = ast_variable_retrieve(cfg, "general", "format"))) {
ast_copy_string(vmfmts, tmp, sizeof(vmfmts));
}
ast_config_destroy(cfg);
} else
ast_log(LOG_WARNING, "Unable to load " VOICEMAIL_CONFIG " - ODBC defaults will be used\n");
#endif
return ast_register_application(app, directory_exec, synopsis, descrip);
}

0
apps/app_disa.c Executable file → Normal file
View File

3
apps/app_dumpchan.c Executable file → Normal file
View File

@@ -5,7 +5,8 @@
*
* Anthony Minessale <anthmct@yahoo.com>
*
* disclaimed to Digium
* A license has been granted to Digium (via disclaimer) for the use of
* this code.
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact

0
apps/app_echo.c Executable file → Normal file
View File

0
apps/app_enumlookup.c Executable file → Normal file
View File

0
apps/app_eval.c Executable file → Normal file
View File

0
apps/app_exec.c Executable file → Normal file
View File

54
apps/app_externalivr.c Executable file → Normal file
View File

@@ -31,6 +31,7 @@
#include <string.h>
#include <unistd.h>
#include <errno.h>
#include <signal.h>
#include "asterisk.h"
@@ -43,6 +44,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/linkedlists.h"
#include "asterisk/app.h"
#include "asterisk/options.h"
static const char *tdesc = "External IVR Interface Application";
@@ -93,9 +96,9 @@ static void send_child_event(FILE *handle, const char event, const char *data,
char tmp[256];
if (!data) {
snprintf(tmp, sizeof(tmp), "%c,%10ld", event, time(NULL));
snprintf(tmp, sizeof(tmp), "%c,%10d", event, (int)time(NULL));
} else {
snprintf(tmp, sizeof(tmp), "%c,%10ld,%s", event, time(NULL), data);
snprintf(tmp, sizeof(tmp), "%c,%10d,%s", event, (int)time(NULL), data);
}
fprintf(handle, "%s\n", tmp);
@@ -150,7 +153,7 @@ static int gen_nextfile(struct gen_state *state)
if (state->current) {
file_to_stream = state->current->filename;
} else {
file_to_stream = "silence-10";
file_to_stream = "silence/10";
u->playing_silence = 1;
}
@@ -250,16 +253,19 @@ static int app_exec(struct ast_channel *chan, void *data)
int res = -1;
int gen_active = 0;
int pid;
char *command;
char *argv[32];
int argc = 1;
char *buf;
char *buf, *command;
FILE *child_commands = NULL;
FILE *child_errors = NULL;
FILE *child_events = NULL;
sigset_t fullset, oldset;
LOCAL_USER_ADD(u);
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
AST_LIST_HEAD_INIT(&u->playlist);
AST_LIST_HEAD_INIT(&u->finishlist);
u->abort_current_sound = 0;
@@ -270,11 +276,13 @@ static int app_exec(struct ast_channel *chan, void *data)
}
buf = ast_strdupa(data);
command = strsep(&buf, "|");
memset(argv, 0, sizeof(argv) / sizeof(argv[0]));
argv[0] = command;
while ((argc < 31) && (argv[argc++] = strsep(&buf, "|")));
argv[argc] = NULL;
if (!buf) {
ast_log(LOG_ERROR, "Out of memory!\n");
LOCAL_USER_REMOVE(u);
return -1;
}
argc = ast_app_separate_args(buf, '|', argv, sizeof(argv) / sizeof(argv[0]));
if (pipe(child_stdin)) {
ast_chan_log(LOG_WARNING, chan, "Could not create pipe for child input: %s\n", strerror(errno));
@@ -311,14 +319,20 @@ static int app_exec(struct ast_channel *chan, void *data)
/* child process */
int i;
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
if (option_highpriority)
ast_set_priority(0);
dup2(child_stdin[0], STDIN_FILENO);
dup2(child_stdout[1], STDOUT_FILENO);
dup2(child_stderr[1], STDERR_FILENO);
for (i = STDERR_FILENO + 1; i < 1024; i++)
close(i);
execv(command, argv);
fprintf(stderr, "Failed to execute '%s': %s\n", command, strerror(errno));
exit(1);
execv(argv[0], argv);
fprintf(stderr, "Failed to execute '%s': %s\n", argv[0], strerror(errno));
_exit(1);
} else {
/* parent process */
int child_events_fd = child_stdin[1];
@@ -331,6 +345,8 @@ static int app_exec(struct ast_channel *chan, void *data)
int waitfds[2] = { child_errors_fd, child_commands_fd };
struct ast_channel *rchan;
pthread_sigmask(SIG_SETMASK, &oldset, NULL);
close(child_stdin[0]);
child_stdin[0] = 0;
close(child_stdout[1]);
@@ -343,8 +359,6 @@ static int app_exec(struct ast_channel *chan, void *data)
goto exit;
}
setvbuf(child_events, NULL, _IONBF, 0);
if (!(child_commands = fdopen(child_commands_fd, "r"))) {
ast_chan_log(LOG_WARNING, chan, "Could not open stream for child commands\n");
goto exit;
@@ -355,6 +369,10 @@ static int app_exec(struct ast_channel *chan, void *data)
goto exit;
}
setvbuf(child_events, NULL, _IONBF, 0);
setvbuf(child_commands, NULL, _IONBF, 0);
setvbuf(child_errors, NULL, _IONBF, 0);
res = 0;
while (1) {
@@ -439,7 +457,7 @@ static int app_exec(struct ast_channel *chan, void *data)
continue;
if (input[0] == 'S') {
if (ast_fileexists(&input[2], NULL, NULL) == -1) {
if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
send_child_event(child_events, 'Z', NULL, chan);
strcpy(&input[2], "exception");
@@ -458,7 +476,7 @@ static int app_exec(struct ast_channel *chan, void *data)
AST_LIST_INSERT_TAIL(&u->playlist, entry, list);
AST_LIST_UNLOCK(&u->playlist);
} else if (input[0] == 'A') {
if (ast_fileexists(&input[2], NULL, NULL) == -1) {
if (ast_fileexists(&input[2], NULL, u->chan->language) == -1) {
ast_chan_log(LOG_WARNING, chan, "Unknown file requested '%s'\n", &input[2]);
send_child_event(child_events, 'Z', NULL, chan);
strcpy(&input[2], "exception");

35
apps/app_festival.c Executable file → Normal file
View File

@@ -53,6 +53,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/config.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/options.h"
#define FESTIVAL_CONFIG "festival.conf"
@@ -126,16 +127,26 @@ static int send_waveform_to_fd(char *waveform, int length, int fd) {
#ifdef __PPC__
char c;
#endif
sigset_t fullset, oldset;
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
res = fork();
if (res < 0)
ast_log(LOG_WARNING, "Fork failed\n");
if (res)
if (res) {
pthread_sigmask(SIG_SETMASK, &oldset, NULL);
return res;
}
for (x=0;x<256;x++) {
if (x != fd)
close(x);
}
if (option_highpriority)
ast_set_priority(0);
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
/*IAS */
#ifdef __PPC__
for( x=0; x<length; x+=2)
@@ -175,7 +186,8 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
if (chan->_state != AST_STATE_UP)
ast_answer(chan);
ast_stopstream(chan);
ast_indicate(chan, -1);
owriteformat = chan->writeformat;
res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
@@ -229,11 +241,13 @@ static int send_waveform_to_channel(struct ast_channel *chan, char *waveform, in
myf.f.data = myf.frdata;
if (ast_write(chan, &myf.f) < 0) {
res = -1;
ast_frfree(f);
break;
}
if (res < needed) { /* last frame */
ast_log(LOG_DEBUG, "Last frame\n");
res=0;
ast_frfree(f);
break;
}
} else {
@@ -455,8 +469,23 @@ static int festival_exec(struct ast_channel *chan, void *vdata)
/* This assumes only one waveform will come back, also LP is unlikely */
wave = 0;
do {
int read_data;
for (n=0; n < 3; )
n += read(fd,ack+n,3-n);
{
read_data = read(fd,ack+n,3-n);
/* this avoids falling in infinite loop
* in case that festival server goes down
* */
if ( read_data == -1 )
{
ast_log(LOG_WARNING,"Unable to read from cache/festival fd\n");
close(fd);
ast_config_destroy(cfg);
LOCAL_USER_REMOVE(u);
return -1;
}
n += read_data;
}
ack[3] = '\0';
if (strcmp(ack,"WV\n") == 0) { /* receive a waveform */
ast_log(LOG_DEBUG,"Festival WV command\n");

0
apps/app_flash.c Executable file → Normal file
View File

0
apps/app_forkcdr.c Executable file → Normal file
View File

0
apps/app_getcpeid.c Executable file → Normal file
View File

48
apps/app_groupcount.c Executable file → Normal file
View File

@@ -70,7 +70,8 @@ static int group_count_exec(struct ast_channel *chan, void *data)
if (ast_strlen_zero(group)) {
grp = pbx_builtin_getvar_helper(chan, category);
strncpy(group, grp, sizeof(group) - 1);
if (!ast_strlen_zero(grp))
ast_copy_string(group, grp, sizeof(group));
}
count = ast_app_group_get_count(group, category);
@@ -154,6 +155,11 @@ static int group_check_exec(struct ast_channel *chan, void *data)
deprecation_warning = 1;
}
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "CheckGroup requires an argument(max[@category][|options])\n");
return 0;
}
if (!(parse = ast_strdupa(data))) {
ast_log(LOG_WARNING, "Memory Error!\n");
LOCAL_USER_REMOVE(u);
@@ -195,10 +201,8 @@ static int group_show_channels(int fd, int argc, char *argv[])
{
#define FORMAT_STRING "%-25s %-20s %-20s\n"
struct ast_channel *c = NULL;
int numchans = 0;
struct ast_var_t *current;
struct varshead *headp;
struct ast_group_info *gi = NULL;
regex_t regexbuf;
int havepattern = 0;
@@ -212,26 +216,20 @@ static int group_show_channels(int fd, int argc, char *argv[])
}
ast_cli(fd, FORMAT_STRING, "Channel", "Group", "Category");
while ( (c = ast_channel_walk_locked(c)) != NULL) {
headp=&c->varshead;
AST_LIST_TRAVERSE(headp,current,entries) {
if (!strncmp(ast_var_name(current), GROUP_CATEGORY_PREFIX "_", strlen(GROUP_CATEGORY_PREFIX) + 1)) {
if (!havepattern || !regexec(&regexbuf, ast_var_value(current), 0, NULL, 0)) {
ast_cli(fd, FORMAT_STRING, c->name, ast_var_value(current),
(ast_var_name(current) + strlen(GROUP_CATEGORY_PREFIX) + 1));
numchans++;
}
} else if (!strcmp(ast_var_name(current), GROUP_CATEGORY_PREFIX)) {
if (!havepattern || !regexec(&regexbuf, ast_var_value(current), 0, NULL, 0)) {
ast_cli(fd, FORMAT_STRING, c->name, ast_var_value(current), "(default)");
numchans++;
}
}
ast_app_group_list_lock();
gi = ast_app_group_list_head();
while (gi) {
if (!havepattern || !regexec(&regexbuf, gi->group, 0, NULL, 0)) {
ast_cli(fd, FORMAT_STRING, gi->chan->name, gi->group, (ast_strlen_zero(gi->category) ? "(default)" : gi->category));
numchans++;
}
numchans++;
ast_mutex_unlock(&c->lock);
gi = AST_LIST_NEXT(gi, list);
}
ast_app_group_list_unlock();
if (havepattern)
regfree(&regexbuf);
@@ -257,8 +255,8 @@ static char *group_count_descrip =
" Calculates the group count for the specified group, or uses\n"
"the current channel's group if not specifed (and non-empty).\n"
"Stores result in GROUPCOUNT. \n"
"This application has been deprecated, please use the function\n"
"GroupCount.\n";
"Note: This application has been deprecated, please use the function\n"
"GROUP_COUNT.\n";
static char *group_set_descrip =
"Usage: SetGroup(groupname[@category])\n"
@@ -284,8 +282,8 @@ static char *group_match_count_descrip =
" Calculates the group count for all groups that match the specified\n"
"pattern. Uses standard regular expression matching (see regex(7)).\n"
"Stores result in GROUPCOUNT. Always returns 0.\n"
"This application has been deprecated, please use the function\n"
"GroupMatchCount.\n";
"Note: This application has been deprecated, please use the function\n"
"GROUP_MATCH_COUNT.\n";
static char show_channels_usage[] =
"Usage: group show channels [pattern]\n"

134
apps/app_hasnewvoicemail.c Executable file → Normal file
View File

@@ -47,6 +47,12 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/utils.h"
#include "asterisk/app.h"
#include "asterisk/options.h"
#ifdef USE_ODBC_STORAGE
#include "asterisk/res_odbc.h"
static char odbc_database[80];
static char odbc_table[80];
#endif
static char *tdesc = "Indicator for whether a voice mailbox has messages in a given folder.";
static char *app_hasvoicemail = "HasVoicemail";
@@ -68,7 +74,7 @@ static char *hasnewvoicemail_descrip =
"Assumes folder 'INBOX' if folder is not specified. Optionally sets <varname> to the number of messages\n"
"in that folder.\n"
" The option string may contain zero of the following character:\n"
" 'j' -- jump to priority n+101, if there is new voicemail in tolder 'folder' or INBOX\n"
" 'j' -- jump to priority n+101, if there is new voicemail in folder 'folder' or INBOX\n"
" This application sets the following channel variable upon completion:\n"
" HASVMSTATUS The result of the new voicemail check returned as a text string as follows\n"
" <# of messages in the folder, 0 for NONE>\n";
@@ -77,26 +83,93 @@ STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int hasvoicemail_internal(char *context, char *box, char *folder)
#ifdef USE_ODBC_STORAGE
static int hasvoicemail_internal(const char *context, const char *mailbox, const char *folder)
{
char vmpath[256];
DIR *vmdir;
struct dirent *vment;
int count=0;
int nummsgs = 0;
int res;
SQLHSTMT stmt;
char sql[256];
char rowdata[20];
snprintf(vmpath,sizeof(vmpath), "%s/voicemail/%s/%s/%s", (char *)ast_config_AST_SPOOL_DIR, context, box, folder);
if ((vmdir = opendir(vmpath))) {
/* No matter what the format of VM, there will always be a .txt file for each message. */
while ((vment = readdir(vmdir))) {
if (!strncmp(vment->d_name + 7, ".txt", 4)) {
count++;
break;
}
if (!folder)
folder = "INBOX";
/* If no mailbox, return immediately */
if (ast_strlen_zero(mailbox))
return 0;
if (ast_strlen_zero(context))
context = "default";
odbc_obj *obj;
obj = fetch_odbc_obj(odbc_database, 0);
if (obj) {
res = SQLAllocHandle(SQL_HANDLE_STMT, obj->con, &stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Alloc Handle failed!\n");
goto yuck;
}
closedir(vmdir);
snprintf(sql, sizeof(sql), "SELECT COUNT(*) FROM %s WHERE dir = '%s/voicemail/%s/%s/%s'", odbc_table, ast_config_AST_SPOOL_DIR, context, mailbox, folder);
res = SQLPrepare(stmt, sql, SQL_NTS);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Prepare failed![%s]\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
goto yuck;
}
res = odbc_smart_execute(obj, stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Execute error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
goto yuck;
}
res = SQLFetch(stmt);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Fetch error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
goto yuck;
}
res = SQLGetData(stmt, 1, SQL_CHAR, rowdata, sizeof(rowdata), NULL);
if ((res != SQL_SUCCESS) && (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, "SQL Get Data error!\n[%s]\n\n", sql);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
goto yuck;
}
nummsgs = atoi(rowdata);
SQLFreeHandle(SQL_HANDLE_STMT, stmt);
} else
ast_log(LOG_WARNING, "Failed to obtain database object for '%s'!\n", odbc_database);
yuck:
return nummsgs;
}
#else
static int hasvoicemail_internal(const char *context, const char *mailbox, const char *folder)
{
DIR *dir;
struct dirent *de;
char fn[256];
int count = 0;
if (ast_strlen_zero(folder))
folder = "INBOX";
if (ast_strlen_zero(context))
context = "default";
/* If no mailbox, return immediately */
if (ast_strlen_zero(mailbox))
return 0;
snprintf(fn, sizeof(fn), "%s/voicemail/%s/%s/%s", ast_config_AST_SPOOL_DIR, context, mailbox, folder);
dir = opendir(fn);
if (!dir)
return 0;
while ((de = readdir(dir))) {
if (!strncasecmp(de->d_name, "msg", 3) && !strcasecmp(de->d_name + 8, "txt"))
count++;
}
closedir(dir);
return count;
}
#endif
static int hasvoicemail_exec(struct ast_channel *chan, void *data)
{
@@ -163,7 +236,7 @@ static int hasvoicemail_exec(struct ast_channel *chan, void *data)
if (vmcount > 0) {
/* Branch to the next extension */
if (priority_jump || option_priority_jumping) {
if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
ast_log(LOG_WARNING, "VM box %s@%s has new voicemail, but extension %s, priority %d doesn't exist\n", vmbox, context, chan->exten, chan->priority + 101);
}
}
@@ -187,7 +260,7 @@ static char *acf_vmcount_exec(struct ast_channel *chan, char *cmd, char *data, c
args = ast_strdupa(data);
if (!args) {
ast_log(LOG_ERROR, "Out of memory");
ast_log(LOG_ERROR, "Out of memory\n");
LOCAL_USER_REMOVE(u);
return buf;
}
@@ -223,6 +296,31 @@ struct ast_custom_function acf_vmcount = {
.read = acf_vmcount_exec,
};
static int load_config(void)
{
#ifdef USE_ODBC_STORAGE
struct ast_config *cfg;
char *tmp;
cfg = ast_config_load("voicemail.conf");
if (cfg) {
if (! (tmp = ast_variable_retrieve(cfg, "general", "odbcstorage")))
tmp = "asterisk";
ast_copy_string(odbc_database, tmp, sizeof(odbc_database));
if (! (tmp = ast_variable_retrieve(cfg, "general", "odbctable")))
tmp = "voicemessages";
ast_copy_string(odbc_table, tmp, sizeof(odbc_table));
ast_config_destroy(cfg);
}
#endif
return 0;
}
int reload(void)
{
return load_config();
}
int unload_module(void)
{
int res;
@@ -239,7 +337,7 @@ int unload_module(void)
int load_module(void)
{
int res;
load_config();
res = ast_custom_function_register(&acf_vmcount);
res |= ast_register_application(app_hasvoicemail, hasvoicemail_exec, hasvoicemail_synopsis, hasvoicemail_descrip);
res |= ast_register_application(app_hasnewvoicemail, hasvoicemail_exec, hasnewvoicemail_synopsis, hasnewvoicemail_descrip);

22
apps/app_ices.c Executable file → Normal file
View File

@@ -44,6 +44,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/options.h"
#define ICES "/usr/bin/ices"
#define LOCAL_ICES "/usr/local/bin/ices"
@@ -67,13 +68,27 @@ static int icesencode(char *filename, int fd)
{
int res;
int x;
sigset_t fullset, oldset;
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
res = fork();
if (res < 0)
ast_log(LOG_WARNING, "Fork failed\n");
if (res)
if (res) {
pthread_sigmask(SIG_SETMASK, &oldset, NULL);
return res;
}
/* Stop ignoring PIPE */
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
if (option_highpriority)
ast_set_priority(0);
dup2(fd, STDIN_FILENO);
for (x=STDERR_FILENO + 1;x<256;x++) {
for (x=STDERR_FILENO + 1;x<1024;x++) {
if ((x != STDIN_FILENO) && (x != STDOUT_FILENO))
close(x);
}
@@ -84,7 +99,7 @@ static int icesencode(char *filename, int fd)
/* As a last-ditch effort, try to use PATH */
execlp("ices", "ices", filename, (char *)NULL);
ast_log(LOG_WARNING, "Execute of ices failed\n");
return -1;
_exit(0);
}
static int ices_exec(struct ast_channel *chan, void *data)
@@ -172,6 +187,7 @@ static int ices_exec(struct ast_channel *chan, void *data)
if (errno != EAGAIN) {
ast_log(LOG_WARNING, "Write failed to pipe: %s\n", strerror(errno));
res = -1;
ast_frfree(f);
break;
}
}

0
apps/app_image.c Executable file → Normal file
View File

0
apps/app_intercom.c Executable file → Normal file
View File

0
apps/app_ivrdemo.c Executable file → Normal file
View File

0
apps/app_lookupblacklist.c Executable file → Normal file
View File

0
apps/app_lookupcidname.c Executable file → Normal file
View File

205
apps/app_macro.c Executable file → Normal file
View File

@@ -60,7 +60,18 @@ static char *descrip =
"If you Goto out of the Macro context, the Macro will terminate and control\n"
"will be returned at the location of the Goto.\n"
"If ${MACRO_OFFSET} is set at termination, Macro will attempt to continue\n"
"at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.\n";
"at priority MACRO_OFFSET + N + 1 if such a step exists, and N + 1 otherwise.\n"
"Extensions: While a macro is being executed, it becomes the current context.\n"
" This means that if a hangup occurs, for instance, that the macro\n"
" will be searched for an 'h' extension, NOT the context from which\n"
" the macro was called. So, make sure to define all appropriate\n"
" extensions in your macro! (you can use 'catch' in AEL) \n"
"WARNING: Because of the way Macro is implemented (it executes the priorities\n"
" contained within it via sub-engine), and a fixed per-thread\n"
" memory stack allowance, macros are limited to 7 levels\n"
" of nesting (macro calling macro calling macro, etc.); It\n"
" may be possible that stack-intensive applications in deeply nested\n"
" macros could cause asterisk to crash earlier than this limit.\n";
static char *if_descrip =
" MacroIf(<expr>?macroname_a[|arg1][:macroname_b[|arg1]])\n"
@@ -87,6 +98,41 @@ STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static struct ast_exten *find_matching_priority(struct ast_context *c, const char *exten, int priority, const char *callerid)
{
struct ast_exten *e;
struct ast_include *i;
struct ast_context *c2;
for (e=ast_walk_context_extensions(c, NULL); e; e=ast_walk_context_extensions(c, e)) {
if (ast_extension_match(ast_get_extension_name(e), exten)) {
int needmatch = ast_get_extension_matchcid(e);
if ((needmatch && ast_extension_match(ast_get_extension_cidmatch(e), callerid)) ||
(!needmatch)) {
/* This is the matching extension we want */
struct ast_exten *p;
for (p=ast_walk_extension_priorities(e, NULL); p; p=ast_walk_extension_priorities(e, p)) {
if (priority != ast_get_extension_priority(p))
continue;
return p;
}
}
}
}
/* No match; run through includes */
for (i=ast_walk_context_includes(c, NULL); i; i=ast_walk_context_includes(c, i)) {
for (c2=ast_walk_contexts(NULL); c2; c2=ast_walk_contexts(c2)) {
if (!strcmp(ast_get_context_name(c2), ast_get_include_name(i))) {
e = find_matching_priority(c2, exten, priority, callerid);
if (e)
return e;
}
}
}
return NULL;
}
static int macro_exec(struct ast_channel *chan, void *data)
{
char *tmp;
@@ -94,17 +140,18 @@ static int macro_exec(struct ast_channel *chan, void *data)
char *macro;
char fullmacro[80];
char varname[80];
char runningapp[80], runningdata[1024];
char *oldargs[MAX_ARGS + 1] = { NULL, };
int argc, x;
int res=0;
char oldexten[256]="";
int oldpriority;
int oldpriority, gosub_level = 0;
char pc[80], depthc[12];
char oldcontext[AST_MAX_CONTEXT] = "";
char *offsets;
int offset, depth;
char *offsets, *s, *inhangupc;
int offset, depth = 0, maxdepth = 7;
int setmacrocontext=0;
int autoloopflag;
int autoloopflag, dead = 0, inhangup = 0;
char *save_macro_exten;
char *save_macro_context;
@@ -119,6 +166,11 @@ static int macro_exec(struct ast_channel *chan, void *data)
LOCAL_USER_ADD(u);
/* does the user want a deeper rabbit hole? */
s = pbx_builtin_getvar_helper(chan, "MACRO_RECURSION");
if (s)
sscanf(s, "%d", &maxdepth);
/* Count how many levels deep the rabbit hole goes */
tmp = pbx_builtin_getvar_helper(chan, "MACRO_DEPTH");
if (tmp) {
@@ -127,7 +179,14 @@ static int macro_exec(struct ast_channel *chan, void *data)
depth = 0;
}
if (depth >= 7) {
/* Used for detecting whether to return when a Macro is called from another Macro after hangup */
if (strcmp(chan->exten, "h") == 0)
pbx_builtin_setvar_helper(chan, "MACRO_IN_HANGUP", "1");
inhangupc = pbx_builtin_getvar_helper(chan, "MACRO_IN_HANGUP");
if (!ast_strlen_zero(inhangupc))
sscanf(inhangupc, "%d", &inhangup);
if (depth >= maxdepth) {
ast_log(LOG_ERROR, "Macro(): possible infinite loop detected. Returning early.\n");
LOCAL_USER_REMOVE(u);
return 0;
@@ -205,8 +264,36 @@ static int macro_exec(struct ast_channel *chan, void *data)
autoloopflag = ast_test_flag(chan, AST_FLAG_IN_AUTOLOOP);
ast_set_flag(chan, AST_FLAG_IN_AUTOLOOP);
while(ast_exists_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num)) {
struct ast_context *c;
struct ast_exten *e;
runningapp[0] = '\0';
runningdata[0] = '\0';
/* What application will execute? */
if (ast_lock_contexts()) {
ast_log(LOG_WARNING, "Failed to lock contexts list\n");
} else {
for (c = ast_walk_contexts(NULL), e = NULL; c; c = ast_walk_contexts(c)) {
if (!strcmp(ast_get_context_name(c), chan->context)) {
if (ast_lock_context(c)) {
ast_log(LOG_WARNING, "Unable to lock context?\n");
} else {
e = find_matching_priority(c, chan->exten, chan->priority, chan->cid.cid_num);
if (e) { /* This will only be undefined for pbx_realtime, which is majorly broken. */
ast_copy_string(runningapp, ast_get_extension_app(e), sizeof(runningapp));
ast_copy_string(runningdata, ast_get_extension_app_data(e), sizeof(runningdata));
}
ast_unlock_context(c);
}
break;
}
}
}
ast_unlock_contexts();
/* Reset the macro depth, if it was changed in the last iteration */
pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
if ((res = ast_spawn_extension(chan, chan->context, chan->exten, chan->priority, chan->cid.cid_num))) {
/* Something bad happened, or a hangup has been requested. */
if (((res >= '0') && (res <= '9')) || ((res >= 'A') && (res <= 'F')) ||
@@ -216,33 +303,91 @@ static int macro_exec(struct ast_channel *chan, void *data)
break;
}
switch(res) {
case MACRO_EXIT_RESULT:
res = 0;
case MACRO_EXIT_RESULT:
res = 0;
goto out;
case AST_PBX_KEEPALIVE:
if (option_debug)
ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited KEEPALIVE in macro %s on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
else if (option_verbose > 1)
if (option_verbose > 1)
ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited KEEPALIVE in macro '%s' on '%s'\n", chan->context, chan->exten, chan->priority, macro, chan->name);
goto out;
break;
default:
if (option_debug)
ast_log(LOG_DEBUG, "Spawn extension (%s,%s,%d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
else if (option_verbose > 1)
if (option_verbose > 1)
ast_verbose( VERBOSE_PREFIX_2 "Spawn extension (%s, %s, %d) exited non-zero on '%s' in macro '%s'\n", chan->context, chan->exten, chan->priority, chan->name, macro);
dead = 1;
goto out;
}
}
if (strcasecmp(chan->context, fullmacro)) {
ast_log(LOG_DEBUG, "Executed application: %s\n", runningapp);
if (!strcasecmp(runningapp, "GOSUB")) {
gosub_level++;
ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
} else if (!strcasecmp(runningapp, "GOSUBIF")) {
char tmp2[1024] = "", *cond, *app, *app2 = tmp2;
pbx_substitute_variables_helper(chan, runningdata, tmp2, sizeof(tmp2) - 1);
cond = strsep(&app2, "?");
app = strsep(&app2, ":");
if (pbx_checkcondition(cond)) {
if (!ast_strlen_zero(app)) {
gosub_level++;
ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
}
} else {
if (!ast_strlen_zero(app2)) {
gosub_level++;
ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
}
}
} else if (!strcasecmp(runningapp, "RETURN")) {
gosub_level--;
ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
} else if (!strcasecmp(runningapp, "STACKPOP")) {
gosub_level--;
ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
} else if (!strncasecmp(runningapp, "EXEC", 4)) {
/* Must evaluate args to find actual app */
char tmp2[1024] = "", *tmp3 = NULL;
pbx_substitute_variables_helper(chan, runningdata, tmp2, sizeof(tmp2) - 1);
if (!strcasecmp(runningapp, "EXECIF")) {
tmp3 = strchr(tmp2, '|');
if (tmp3)
*tmp3++ = '\0';
if (!pbx_checkcondition(tmp2))
tmp3 = NULL;
} else
tmp3 = tmp2;
if (tmp3)
ast_log(LOG_DEBUG, "Last app: %s\n", tmp3);
if (tmp3 && !strncasecmp(tmp3, "GOSUB", 5)) {
gosub_level++;
ast_log(LOG_DEBUG, "Incrementing gosub_level\n");
} else if (tmp3 && !strncasecmp(tmp3, "RETURN", 6)) {
gosub_level--;
ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
} else if (tmp3 && !strncasecmp(tmp3, "STACKPOP", 8)) {
gosub_level--;
ast_log(LOG_DEBUG, "Decrementing gosub_level\n");
}
}
if (gosub_level == 0 && strcasecmp(chan->context, fullmacro)) {
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "Channel '%s' jumping out of macro '%s'\n", chan->name, macro);
break;
}
/* don't stop executing extensions when we're in "h" */
if (chan->_softhangup && strcasecmp(oldexten,"h")) {
ast_log(LOG_DEBUG, "Extension %s, priority %d returned normally even though call was hung up\n",
chan->exten, chan->priority);
if (chan->_softhangup && !inhangup) {
ast_log(LOG_DEBUG, "Extension %s, macroexten %s, priority %d returned normally even though call was hung up\n",
chan->exten, chan->macroexten, chan->priority);
goto out;
}
chan->priority++;
@@ -250,37 +395,44 @@ static int macro_exec(struct ast_channel *chan, void *data)
out:
/* Reset the depth back to what it was when the routine was entered (like if we called Macro recursively) */
snprintf(depthc, sizeof(depthc), "%d", depth);
pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
if (!dead) {
pbx_builtin_setvar_helper(chan, "MACRO_DEPTH", depthc);
ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
for (x=1; x<argc; x++) {
ast_set2_flag(chan, autoloopflag, AST_FLAG_IN_AUTOLOOP);
}
for (x = 1; x < argc; x++) {
/* Restore old arguments and delete ours */
snprintf(varname, sizeof(varname), "ARG%d", x);
if (oldargs[x]) {
pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
if (!dead)
pbx_builtin_setvar_helper(chan, varname, oldargs[x]);
free(oldargs[x]);
} else {
} else if (!dead) {
pbx_builtin_setvar_helper(chan, varname, NULL);
}
}
/* Restore macro variables */
pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
if (!dead) {
pbx_builtin_setvar_helper(chan, "MACRO_EXTEN", save_macro_exten);
pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
}
if (save_macro_exten)
free(save_macro_exten);
pbx_builtin_setvar_helper(chan, "MACRO_CONTEXT", save_macro_context);
if (save_macro_context)
free(save_macro_context);
pbx_builtin_setvar_helper(chan, "MACRO_PRIORITY", save_macro_priority);
if (save_macro_priority)
free(save_macro_priority);
if (setmacrocontext) {
if (!dead && setmacrocontext) {
chan->macrocontext[0] = '\0';
chan->macroexten[0] = '\0';
chan->macropriority = 0;
}
if (!strcasecmp(chan->context, fullmacro)) {
if (!dead && !strcasecmp(chan->context, fullmacro)) {
/* If we're leaving the macro normally, restore original information */
chan->priority = oldpriority;
ast_copy_string(chan->context, oldcontext, sizeof(chan->context));
@@ -299,7 +451,8 @@ static int macro_exec(struct ast_channel *chan, void *data)
}
}
pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
if (!dead)
pbx_builtin_setvar_helper(chan, "MACRO_OFFSET", save_macro_offset);
if (save_macro_offset)
free(save_macro_offset);
LOCAL_USER_REMOVE(u);
@@ -328,7 +481,7 @@ static int macroif_exec(struct ast_channel *chan, void *data)
*label_b = '\0';
label_b++;
}
if (ast_true(expr))
if (pbx_checkcondition(expr))
macro_exec(chan, label_a);
else if (label_b)
macro_exec(chan, label_b);

3
apps/app_math.c Executable file → Normal file
View File

@@ -69,7 +69,8 @@ static char *math_descrip =
"Perform floating point calculation on number 1 to number 2 and \n"
"store the result in returnvar. Valid ops are: \n"
" +,-,/,*,%,<,>,>=,<=,==\n"
"and behave as their C equivalents.\n";
"and behave as their C equivalents.\n"
" This application has been deprecated in favor of the MATH function.\n";
#define ADDFUNCTION 0
#define DIVIDEFUNCTION 1

4
apps/app_md5.c Executable file → Normal file
View File

@@ -157,9 +157,9 @@ static int md5check_exec(struct ast_channel *chan, void *data)
ast_log(LOG_DEBUG, "ERROR: MD5 not verified: %s -- %s\n", args.md5hash, args.string);
pbx_builtin_setvar_helper(chan, "CHECKMD5STATUS", "NOMATCH");
if (priority_jump || option_priority_jumping) {
if (!ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
if (ast_goto_if_exists(chan, chan->context, chan->exten, chan->priority + 101))
if (option_debug > 2)
ast_log(LOG_DEBUG, "ERROR: Can't jump to exten+101 (e%s,p%d), sorry\n", chan->exten,chan->priority+101);
ast_log(LOG_DEBUG, "Can't jump to exten+101 (e%s,p%d), sorry\n", chan->exten,chan->priority+101);
}
LOCAL_USER_REMOVE(u);
return res;

522
apps/app_meetme.c Executable file → Normal file

File diff suppressed because it is too large Load Diff

15
apps/app_milliwatt.c Executable file → Normal file
View File

@@ -73,30 +73,29 @@ static void milliwatt_release(struct ast_channel *chan, void *data)
static int milliwatt_generate(struct ast_channel *chan, void *data, int len, int samples)
{
struct ast_frame wf;
unsigned char waste[AST_FRIENDLY_OFFSET];
unsigned char buf[640];
unsigned char buf[AST_FRIENDLY_OFFSET + 640];
int i,*indexp = (int *) data;
if (len > sizeof(buf))
if (len + AST_FRIENDLY_OFFSET > sizeof(buf))
{
ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",(int)sizeof(buf),len);
len = sizeof(buf);
ast_log(LOG_WARNING,"Only doing %d bytes (%d bytes requested)\n",(int)(sizeof(buf) - AST_FRIENDLY_OFFSET),len);
len = sizeof(buf) - AST_FRIENDLY_OFFSET;
}
waste[0] = 0; /* make compiler happy */
wf.frametype = AST_FRAME_VOICE;
wf.subclass = AST_FORMAT_ULAW;
wf.offset = AST_FRIENDLY_OFFSET;
wf.mallocd = 0;
wf.data = buf;
wf.data = buf + AST_FRIENDLY_OFFSET;
wf.datalen = len;
wf.samples = wf.datalen;
wf.src = "app_milliwatt";
wf.delivery.tv_sec = 0;
wf.delivery.tv_usec = 0;
wf.prev = wf.next = NULL;
/* create a buffer containing the digital milliwatt pattern */
for(i = 0; i < len; i++)
{
buf[i] = digital_milliwatt[(*indexp)++];
buf[AST_FRIENDLY_OFFSET + i] = digital_milliwatt[(*indexp)++];
*indexp &= 7;
}
if (ast_write(chan,&wf) < 0)

222
apps/app_mixmonitor.c Executable file → Normal file
View File

@@ -61,14 +61,15 @@ static const char *desc = ""
"Valid options:\n"
" a - Append to the file instead of overwriting it.\n"
" b - Only save audio to the file while the channel is bridged.\n"
" Note: does not include conferences.\n"
" Note: Does not include conferences or sounds played to each bridged\n"
" party.\n"
" v(<x>) - Adjust the heard volume by a factor of <x> (range -4 to 4)\n"
" V(<x>) - Adjust the spoken volume by a factor of <x> (range -4 to 4)\n"
" W(<x>) - Adjust the both heard and spoken volumes by a factor of <x>\n"
" (range -4 to 4)\n\n"
"<command> will be executed when the recording is over\n"
"Any strings matching ^{X} will be unescaped to ${X} and \n"
"all variables will be evaluated at that time.\n"
"Any strings matching ^{X} will be unescaped to ${X}.\n"
"All variables will be evaluated at the time MixMonitor is called.\n"
"The variable MIXMONITOR_FILENAME will contain the filename used to record.\n"
"";
@@ -79,12 +80,11 @@ LOCAL_USER_DECL;
static const char *mixmonitor_spy_type = "MixMonitor";
struct mixmonitor {
struct ast_channel *chan;
struct ast_channel_spy spy;
char *filename;
char *post_process;
char *name;
unsigned int flags;
int readvol;
int writevol;
};
enum {
@@ -110,21 +110,6 @@ AST_APP_OPTIONS(mixmonitor_opts, {
AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
});
static void stopmon(struct ast_channel *chan, struct ast_channel_spy *spy)
{
/* If our status has changed, then the channel we're spying on is gone....
DON'T TOUCH IT!!! RUN AWAY!!! */
if (spy->status != CHANSPY_RUNNING)
return;
if (!chan)
return;
ast_mutex_lock(&chan->lock);
ast_channel_spy_remove(chan, spy);
ast_mutex_unlock(&chan->lock);
}
static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy)
{
struct ast_channel *peer;
@@ -136,7 +121,7 @@ static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy)
ast_mutex_lock(&chan->lock);
res = ast_channel_spy_add(chan, spy);
ast_mutex_unlock(&chan->lock);
if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);
@@ -148,124 +133,83 @@ static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy)
static void *mixmonitor_thread(void *obj)
{
struct mixmonitor *mixmonitor = obj;
struct ast_channel_spy spy;
struct ast_frame *f = NULL;
struct ast_filestream *fs = NULL;
char *ext, *name;
unsigned int oflags;
struct ast_frame *f;
char post_process[1024] = "";
char *ext;
int errflag = 0;
STANDARD_INCREMENT_USECOUNT;
name = ast_strdupa(mixmonitor->chan->name);
oflags = O_CREAT|O_WRONLY;
oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
if ((ext = strrchr(mixmonitor->filename, '.'))) {
*(ext++) = '\0';
} else {
ext = "raw";
}
fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644);
if (!fs) {
ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
goto out;
}
if (ast_test_flag(mixmonitor, MUXFLAG_APPEND))
ast_seekstream(fs, 0, SEEK_END);
memset(&spy, 0, sizeof(spy));
ast_set_flag(&spy, CHANSPY_FORMAT_AUDIO);
ast_set_flag(&spy, CHANSPY_MIXAUDIO);
spy.type = mixmonitor_spy_type;
spy.status = CHANSPY_RUNNING;
spy.read_queue.format = AST_FORMAT_SLINEAR;
spy.write_queue.format = AST_FORMAT_SLINEAR;
if (mixmonitor->readvol) {
ast_set_flag(&spy, CHANSPY_READ_VOLADJUST);
spy.read_vol_adjustment = mixmonitor->readvol;
}
if (mixmonitor->writevol) {
ast_set_flag(&spy, CHANSPY_WRITE_VOLADJUST);
spy.write_vol_adjustment = mixmonitor->writevol;
}
ast_mutex_init(&spy.lock);
if (startmon(mixmonitor->chan, &spy)) {
ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
spy.type, mixmonitor->chan->name);
goto out2;
}
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", name);
ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", mixmonitor->name);
while (1) {
ast_mutex_lock(&mixmonitor->spy.lock);
while (mixmonitor->spy.chan) {
struct ast_frame *next;
int write;
ast_mutex_lock(&spy.lock);
ast_channel_spy_trigger_wait(&spy);
ast_channel_spy_trigger_wait(&mixmonitor->spy);
if (ast_check_hangup(mixmonitor->chan) || spy.status != CHANSPY_RUNNING) {
ast_mutex_unlock(&spy.lock);
if (!mixmonitor->spy.chan || mixmonitor->spy.status != CHANSPY_RUNNING)
break;
}
while (1) {
if (!(f = ast_channel_spy_read_frame(&spy, SAMPLES_PER_FRAME)))
if (!(f = ast_channel_spy_read_frame(&mixmonitor->spy, SAMPLES_PER_FRAME)))
break;
write = (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) ||
ast_bridged_channel(mixmonitor->chan));
ast_bridged_channel(mixmonitor->spy.chan));
/* it is possible for ast_channel_spy_read_frame() to return a chain
of frames if a queue flush was necessary, so process them
*/
for (; f; f = next) {
next = f->next;
if (write)
ast_writestream(fs, f);
if (write && errflag == 0) {
if (!fs) {
/* Determine creation flags and filename plus extension for filestream */
oflags = O_CREAT | O_WRONLY;
oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
if ((ext = strrchr(mixmonitor->filename, '.')))
*(ext++) = '\0';
else
ext = "raw";
/* Move onto actually creating the filestream */
if (!(fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) {
ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
errflag = 1;
}
}
if (fs)
ast_writestream(fs, f);
}
ast_frfree(f);
}
}
ast_mutex_unlock(&spy.lock);
}
ast_mutex_unlock(&mixmonitor->spy.lock);
if (mixmonitor->post_process) {
char *p;
for (p = mixmonitor->post_process; *p ; p++) {
if (*p == '^' && *(p+1) == '{') {
*p = '$';
}
}
pbx_substitute_variables_helper(mixmonitor->chan, mixmonitor->post_process, post_process, sizeof(post_process) - 1);
}
stopmon(mixmonitor->chan, &spy);
ast_channel_spy_free(&mixmonitor->spy);
if (option_verbose > 1)
ast_verbose(VERBOSE_PREFIX_2 "End MixMonitor Recording %s\n", name);
ast_verbose(VERBOSE_PREFIX_2 "End MixMonitor Recording %s\n", mixmonitor->name);
if (!ast_strlen_zero(post_process)) {
if (mixmonitor->post_process) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_2 "Executing [%s]\n", post_process);
ast_safe_system(post_process);
ast_verbose(VERBOSE_PREFIX_2 "Executing [%s]\n", mixmonitor->post_process);
ast_safe_system(mixmonitor->post_process);
}
out2:
ast_mutex_destroy(&spy.lock);
if (fs)
ast_closestream(fs);
out:
free(mixmonitor);
STANDARD_DECREMENT_USECOUNT;
@@ -279,32 +223,76 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
pthread_attr_t attr;
pthread_t thread;
struct mixmonitor *mixmonitor;
int len;
char postprocess2[1024] = "";
size_t len;
len = sizeof(*mixmonitor) + strlen(filename) + 1;
if (!ast_strlen_zero(post_process))
len += strlen(post_process) + 1;
len = sizeof(*mixmonitor) + strlen(chan->name) + strlen(filename) + 2;
/* If a post process system command is given attach it to the structure */
if (!ast_strlen_zero(post_process)) {
char *p1, *p2;
p1 = ast_strdupa(post_process);
for (p2 = p1; *p2 ; p2++) {
if (*p2 == '^' && *(p2+1) == '{') {
*p2 = '$';
}
}
pbx_substitute_variables_helper(chan, p1, postprocess2, sizeof(postprocess2) - 1);
if (!ast_strlen_zero(postprocess2))
len += strlen(postprocess2) + 1;
}
/* Pre-allocate mixmonitor structure and spy */
if (!(mixmonitor = calloc(1, len))) {
ast_log(LOG_ERROR, "Memory Error!\n");
return;
}
mixmonitor->chan = chan;
mixmonitor->filename = (char *) mixmonitor + sizeof(*mixmonitor);
strcpy(mixmonitor->filename, filename);
if (!ast_strlen_zero(post_process)) {
mixmonitor->post_process = mixmonitor->filename + strlen(filename) + 1;
strcpy(mixmonitor->post_process, post_process);
}
mixmonitor->readvol = readvol;
mixmonitor->writevol = writevol;
/* Copy over flags and channel name */
mixmonitor->flags = flags;
mixmonitor->name = (char *) mixmonitor + sizeof(*mixmonitor);
strcpy(mixmonitor->name, chan->name);
if (!ast_strlen_zero(postprocess2)) {
mixmonitor->post_process = mixmonitor->name + strlen(mixmonitor->name) + strlen(filename) + 2;
strcpy(mixmonitor->post_process, postprocess2);
}
mixmonitor->filename = (char *) mixmonitor + sizeof(*mixmonitor) + strlen(chan->name) + 1;
strcpy(mixmonitor->filename, filename);
/* Setup the actual spy before creating our thread */
ast_set_flag(&mixmonitor->spy, CHANSPY_FORMAT_AUDIO);
ast_set_flag(&mixmonitor->spy, CHANSPY_MIXAUDIO);
mixmonitor->spy.type = mixmonitor_spy_type;
mixmonitor->spy.status = CHANSPY_RUNNING;
mixmonitor->spy.read_queue.format = AST_FORMAT_SLINEAR;
mixmonitor->spy.write_queue.format = AST_FORMAT_SLINEAR;
if (readvol) {
ast_set_flag(&mixmonitor->spy, CHANSPY_READ_VOLADJUST);
mixmonitor->spy.read_vol_adjustment = readvol;
}
if (writevol) {
ast_set_flag(&mixmonitor->spy, CHANSPY_WRITE_VOLADJUST);
mixmonitor->spy.write_vol_adjustment = writevol;
}
ast_mutex_init(&mixmonitor->spy.lock);
if (startmon(chan, &mixmonitor->spy)) {
ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
mixmonitor->spy.type, chan->name);
/* Since we couldn't add ourselves - bail out! */
ast_mutex_destroy(&mixmonitor->spy.lock);
free(mixmonitor);
return;
}
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
ast_pthread_create(&thread, &attr, mixmonitor_thread, mixmonitor);
pthread_attr_destroy(&attr);
}
static int mixmonitor_exec(struct ast_channel *chan, void *data)

19
apps/app_mp3.c Executable file → Normal file
View File

@@ -43,6 +43,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/options.h"
#define LOCAL_MPG_123 "/usr/local/bin/mpg123"
#define MPG_123 "/usr/bin/mpg123"
@@ -66,13 +67,25 @@ static int mp3play(char *filename, int fd)
{
int res;
int x;
sigset_t fullset, oldset;
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
res = fork();
if (res < 0)
ast_log(LOG_WARNING, "Fork failed\n");
if (res)
if (res) {
pthread_sigmask(SIG_SETMASK, &oldset, NULL);
return res;
}
if (option_highpriority)
ast_set_priority(0);
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
dup2(fd, STDOUT_FILENO);
for (x=0;x<256;x++) {
for (x=STDERR_FILENO + 1;x<256;x++) {
if (x != STDOUT_FILENO)
close(x);
}
@@ -94,7 +107,7 @@ static int mp3play(char *filename, int fd)
execlp("mpg123", "mpg123", "-q", "-s", "-f", "8192", "--mono", "-r", "8000", filename, (char *)NULL);
}
ast_log(LOG_WARNING, "Execute of mpg123 failed\n");
return -1;
_exit(0);
}
static int timed_read(int fd, void *data, int datalen, int timeout)

20
apps/app_nbscat.c Executable file → Normal file
View File

@@ -44,6 +44,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/options.h"
#define LOCAL_NBSCAT "/usr/local/bin/nbscat8k"
#define NBSCAT "/usr/bin/nbscat8k"
@@ -70,13 +71,26 @@ static int NBScatplay(int fd)
{
int res;
int x;
sigset_t fullset, oldset;
sigfillset(&fullset);
pthread_sigmask(SIG_BLOCK, &fullset, &oldset);
res = fork();
if (res < 0)
ast_log(LOG_WARNING, "Fork failed\n");
if (res)
if (res) {
pthread_sigmask(SIG_SETMASK, &oldset, NULL);
return res;
}
signal(SIGPIPE, SIG_DFL);
pthread_sigmask(SIG_UNBLOCK, &fullset, NULL);
if (option_highpriority)
ast_set_priority(0);
dup2(fd, STDOUT_FILENO);
for (x=0;x<256;x++) {
for (x = STDERR_FILENO + 1; x < 1024; x++) {
if (x != STDOUT_FILENO)
close(x);
}
@@ -84,7 +98,7 @@ static int NBScatplay(int fd)
execl(NBSCAT, "nbscat8k", "-d", (char *)NULL);
execl(LOCAL_NBSCAT, "nbscat8k", "-d", (char *)NULL);
ast_log(LOG_WARNING, "Execute of nbscat8k failed\n");
return -1;
_exit(0);
}
static int timed_read(int fd, void *data, int datalen)

32
apps/app_osplookup.c Executable file → Normal file
View File

@@ -216,27 +216,33 @@ static int ospnext_exec(struct ast_channel *chan, void *data)
cause = str2cause(args.cause);
temp = pbx_builtin_getvar_helper(chan, "OSPHANDLE");
result.handle = -1;
if (!ast_strlen_zero(temp) && (sscanf(temp, "%d", &result.handle) == 1) && (result.handle > -1)) {
if ((res = ast_osp_next(&result, cause)) > 0) {
char tmp[80];
snprintf(tmp, sizeof(tmp), "%d", result.handle);
pbx_builtin_setvar_helper(chan, "_OSPHANDLE", tmp);
pbx_builtin_setvar_helper(chan, "_OSPTECH", result.tech);
pbx_builtin_setvar_helper(chan, "_OSPDEST", result.dest);
pbx_builtin_setvar_helper(chan, "_OSPTOKEN", result.token);
snprintf(tmp, sizeof(tmp), "%d", result.numresults);
pbx_builtin_setvar_helper(chan, "_OSPRESULTS", tmp);
pbx_builtin_setvar_helper(chan, "OSPNEXTSTATUS", "SUCCESS");
}
if (ast_strlen_zero(temp) || (sscanf(temp, "%d", &result.handle) != 1)) {
result.handle = -1;
}
temp = pbx_builtin_getvar_helper(chan, "OSPRESULTS");
if (ast_strlen_zero(temp) || (sscanf(temp, "%d", &result.numresults) != 1)) {
result.numresults = 0;
}
if ((res = ast_osp_next(&result, cause)) > 0) {
char tmp[80];
snprintf(tmp, sizeof(tmp), "%d", result.handle);
pbx_builtin_setvar_helper(chan, "_OSPHANDLE", tmp);
pbx_builtin_setvar_helper(chan, "_OSPTECH", result.tech);
pbx_builtin_setvar_helper(chan, "_OSPDEST", result.dest);
pbx_builtin_setvar_helper(chan, "_OSPTOKEN", result.token);
snprintf(tmp, sizeof(tmp), "%d", result.numresults);
pbx_builtin_setvar_helper(chan, "_OSPRESULTS", tmp);
pbx_builtin_setvar_helper(chan, "OSPNEXTSTATUS", "SUCCESS");
} else {
if (!res) {
if (result.handle < 0)
ast_log(LOG_NOTICE, "OSP Lookup Next failed for handle '%d'\n", result.handle);
else
ast_log(LOG_DEBUG, "No OSP handle specified\n");
pbx_builtin_setvar_helper(chan, "OSPNEXTSTATUS", "FAILED");
} else
ast_log(LOG_DEBUG, "Got hangup on '%s' while doing OSP Next!\n", chan->name);
pbx_builtin_setvar_helper(chan, "OSPNEXTSTATUS", "FAILED");
}
if (!res) {
/* Look for a "busy" place */

48
apps/app_page.c Executable file → Normal file
View File

@@ -1,7 +1,7 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (c) 2004 - 2005 Digium, Inc. All rights reserved.
* Copyright (c) 2004 - 2006 Digium, Inc. All rights reserved.
*
* Mark Spencer <markster@digium.com>
*
@@ -40,6 +40,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/module.h"
#include "asterisk/file.h"
#include "asterisk/app.h"
#include "asterisk/chanvars.h"
static const char *tdesc = "Page Multiple Phones";
@@ -55,7 +56,7 @@ static const char *page_descrip =
"caller is dumped into the conference as a speaker and the room is\n"
"destroyed when the original caller leaves. Valid options are:\n"
" d - full duplex audio\n"
" q - quiet, do not play beep to caller\n";
" q - quiet, do not play beep to caller\n";
STANDARD_LOCAL_USER;
@@ -77,13 +78,14 @@ struct calloutdata {
char tech[64];
char resource[256];
char meetmeopts[64];
struct ast_variable *variables;
};
static void *page_thread(void *data)
{
struct calloutdata *cd = data;
ast_pbx_outgoing_app(cd->tech, AST_FORMAT_SLINEAR, cd->resource, 30000,
"MeetMe", cd->meetmeopts, NULL, 0, cd->cidnum, cd->cidname, NULL, NULL);
"MeetMe", cd->meetmeopts, NULL, 0, cd->cidnum, cd->cidname, cd->variables, NULL, NULL);
free(cd);
return NULL;
}
@@ -91,6 +93,9 @@ static void *page_thread(void *data)
static void launch_page(struct ast_channel *chan, const char *meetmeopts, const char *tech, const char *resource)
{
struct calloutdata *cd;
const char *varname;
struct ast_variable *lastvar = NULL;
struct ast_var_t *varptr;
pthread_t t;
pthread_attr_t attr;
cd = malloc(sizeof(struct calloutdata));
@@ -101,12 +106,36 @@ static void launch_page(struct ast_channel *chan, const char *meetmeopts, const
ast_copy_string(cd->tech, tech, sizeof(cd->tech));
ast_copy_string(cd->resource, resource, sizeof(cd->resource));
ast_copy_string(cd->meetmeopts, meetmeopts, sizeof(cd->meetmeopts));
AST_LIST_TRAVERSE(&chan->varshead, varptr, entries) {
if (!(varname = ast_var_full_name(varptr)))
continue;
if (varname[0] == '_') {
struct ast_variable *newvar = NULL;
if (varname[1] == '_') {
newvar = ast_variable_new(varname, ast_var_value(varptr));
} else {
newvar = ast_variable_new(&varname[1], ast_var_value(varptr));
}
if (newvar) {
if (lastvar)
lastvar->next = newvar;
else
cd->variables = newvar;
lastvar = newvar;
}
}
}
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
if (ast_pthread_create(&t, &attr, page_thread, cd)) {
ast_log(LOG_WARNING, "Unable to create paging thread: %s\n", strerror(errno));
free(cd);
}
pthread_attr_destroy(&attr);
}
}
@@ -121,6 +150,7 @@ static int page_exec(struct ast_channel *chan, void *data)
struct ast_app *app;
char *tmp;
int res=0;
char originator[AST_CHANNEL_NAME];
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
@@ -142,12 +172,21 @@ static int page_exec(struct ast_channel *chan, void *data)
return -1;
}
ast_copy_string(originator, chan->name, sizeof(originator));
if ((tmp = strchr(originator, '-')))
*tmp = '\0';
tmp = strsep(&options, "|");
if (options)
ast_app_parse_options(page_opts, &flags, NULL, options);
snprintf(meetmeopts, sizeof(meetmeopts), "%ud|%sqxdw", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m");
snprintf(meetmeopts, sizeof(meetmeopts), "%ud|%sqxdw(5)", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m");
while ((tech = strsep(&tmp, "&"))) {
/* don't call the originating device */
if (!strcasecmp(tech, originator))
continue;
if ((resource = strchr(tech, '/'))) {
*resource++ = '\0';
launch_page(chan, meetmeopts, tech, resource);
@@ -155,6 +194,7 @@ static int page_exec(struct ast_channel *chan, void *data)
ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
}
}
if (!ast_test_flag(&flags, PAGE_QUIET)) {
res = ast_streamfile(chan, "beep", chan->language);
if (!res)

0
apps/app_parkandannounce.c Executable file → Normal file
View File

9
apps/app_playback.c Executable file → Normal file
View File

@@ -110,6 +110,7 @@ static int playback_exec(struct ast_channel *chan, void *data)
if (chan->_state != AST_STATE_UP) {
if (option_skip) {
/* At the user's option, skip if the line is not up */
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "SUCCESS");
LOCAL_USER_REMOVE(u);
return 0;
} else if (!option_noanswer)
@@ -137,11 +138,11 @@ static int playback_exec(struct ast_channel *chan, void *data)
}
front = back;
}
if (mres)
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "FAILED");
else
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "SUCCESS");
}
if (mres)
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "FAILED");
else
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", "SUCCESS");
LOCAL_USER_REMOVE(u);
return res;
}

16
apps/app_privacy.c Executable file → Normal file
View File

@@ -193,13 +193,21 @@ static int privacy_exec (struct ast_channel *chan, void *data)
}
/*Got a number, play sounds and send them on their way*/
if ((retries < maxretries) && res == 1 ) {
if ((retries < maxretries) && res >= 0 ) {
res = ast_streamfile(chan, "privacy-thankyou", chan->language);
if (!res)
res = ast_waitstream(chan, "");
ast_set_callerid (chan, phone, "Privacy Manager", NULL);
if (option_verbose > 2)
ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s\n",phone);
ast_set_callerid (chan, phone, "Privacy Manager", NULL);
/* Clear the unavailable presence bit so if it came in on PRI
* the caller id will now be passed out to other channels
*/
chan->cid.cid_pres &= (AST_PRES_UNAVAILABLE ^ 0xFF);
if (option_verbose > 2) {
ast_verbose (VERBOSE_PREFIX_3 "Changed Caller*ID to %s, callerpres to %d\n",phone,chan->cid.cid_pres);
}
pbx_builtin_setvar_helper(chan, "PRIVACYMGRSTATUS", "SUCCESS");
} else {
if (priority_jump || option_priority_jumping)

740
apps/app_queue.c Executable file → Normal file

File diff suppressed because it is too large Load Diff

7
apps/app_random.c Executable file → Normal file
View File

@@ -82,7 +82,7 @@ static int random_exec(struct ast_channel *chan, void *data)
if ((!prob) || (sscanf(prob, "%d", &probint) != 1))
probint = 0;
if ((random() % 100) + probint > 100) {
if ((random() % 100) + probint >= 100) {
res = ast_parseable_goto(chan, s);
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Random branches to (%s,%s,%d)\n",
@@ -116,9 +116,8 @@ char *description(void)
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
/* Don't allow unload, since rand(3) depends upon this module being here. */
return 1;
}
char *key()

0
apps/app_read.c Executable file → Normal file
View File

17
apps/app_readfile.c Executable file → Normal file
View File

@@ -95,15 +95,16 @@ static int readfile_exec(struct ast_channel *chan, void *data)
}
}
returnvar = ast_read_textfile(file);
if(len > 0){
if(len < strlen(returnvar))
returnvar[len]='\0';
else
ast_log(LOG_WARNING,"%s is longer than %d, and %d \n", file, len, (int)strlen(returnvar));
if ((returnvar = ast_read_textfile(file))) {
if (len > 0) {
if (len < strlen(returnvar))
returnvar[len]='\0';
else
ast_log(LOG_WARNING, "%s is longer than %d, and %d \n", file, len, (int)strlen(returnvar));
}
pbx_builtin_setvar_helper(chan, varname, returnvar);
free(returnvar);
}
pbx_builtin_setvar_helper(chan, varname, returnvar);
free(returnvar);
LOCAL_USER_REMOVE(u);
return res;
}

0
apps/app_realtime.c Executable file → Normal file
View File

41
apps/app_record.c Executable file → Normal file
View File

@@ -41,6 +41,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/dsp.h"
#include "asterisk/utils.h"
#include "asterisk/options.h"
#include "asterisk/app.h"
static char *tdesc = "Trivial Record Application";
@@ -64,7 +65,8 @@ static char *descrip =
" 't' : use alternate '*' terminator key instead of default '#'\n"
"\n"
"If filename contains '%d', these characters will be replaced with a number\n"
"incremented by one each time the file is recorded. \n\n"
"incremented by one each time the file is recorded. A channel variable\n"
"named RECORDED_FILE will also be set, which contains the final filemname.\n\n"
"Use 'show file formats' to see the available formats on your system\n\n"
"User can press '#' to terminate the recording and continue to the next priority.\n\n"
"If the user should hangup during a recording, all data will be lost and the\n"
@@ -183,8 +185,35 @@ static int record_exec(struct ast_channel *chan, void *data)
/* these are to allow the use of the %d in the config file for a wild card of sort to
create a new file with the inputed name scheme */
if (percentflag) {
AST_DECLARE_APP_ARGS(fname,
AST_APP_ARG(piece)[100];
);
char *tmp2 = ast_strdupa(filename);
char countstring[15];
int i;
/* Separate each piece out by the format specifier */
/* AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%'); */
fname.argc = ast_app_separate_args(tmp2, '%', fname.argv, (sizeof(fname) - sizeof(fname.argc)) / sizeof(fname.argv[0]));
do {
snprintf(tmp, sizeof(tmp), filename, count);
int tmplen;
/* First piece has no leading percent, so it's copied verbatim */
ast_copy_string(tmp, fname.piece[0], sizeof(tmp));
tmplen = strlen(tmp);
for (i = 1; i < fname.argc; i++) {
if (fname.piece[i][0] == 'd') {
/* Substitute the count */
snprintf(countstring, sizeof(countstring), "%d", count);
ast_copy_string(tmp + tmplen, countstring, sizeof(tmp) - tmplen);
tmplen += strlen(countstring);
} else if (tmplen + 2 < sizeof(tmp)) {
/* Unknown format specifier - just copy it verbatim */
tmp[tmplen++] = '%';
tmp[tmplen++] = fname.piece[i][0];
}
/* Copy the remaining portion of the piece */
ast_copy_string(tmp + tmplen, &(fname.piece[i][1]), sizeof(tmp) - tmplen);
}
count++;
} while ( ast_fileexists(tmp, ext, chan->language) != -1 );
pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
@@ -277,6 +306,7 @@ static int record_exec(struct ast_channel *chan, void *data)
if (res) {
ast_log(LOG_WARNING, "Problem writing frame\n");
ast_frfree(f);
break;
}
@@ -295,16 +325,15 @@ static int record_exec(struct ast_channel *chan, void *data)
break;
}
}
}
if (f->frametype == AST_FRAME_VIDEO) {
} else if (f->frametype == AST_FRAME_VIDEO) {
res = ast_writestream(s, f);
if (res) {
ast_log(LOG_WARNING, "Problem writing frame\n");
ast_frfree(f);
break;
}
}
if ((f->frametype == AST_FRAME_DTMF) &&
} else if ((f->frametype == AST_FRAME_DTMF) &&
(f->subclass == terminator)) {
ast_frfree(f);
break;

100
apps/app_rpt.c Executable file → Normal file
View File

@@ -897,55 +897,54 @@ static int telem_lookup(struct ast_channel *chan, char *node, char *name)
static int get_wait_interval(struct rpt *myrpt, int type)
{
int interval;
char *wait_times;
char *wait_times_save;
wait_times_save = NULL;
wait_times = ast_variable_retrieve(cfg, myrpt->name, "wait_times");
if(wait_times){
wait_times_save = ast_strdupa(wait_times);
if(!wait_times_save){
ast_log(LOG_WARNING, "Out of memory in wait_interval()\n");
wait_times = NULL;
}
}
switch(type){
case DLY_TELEM:
if(wait_times)
interval = retrieve_astcfgint(wait_times_save, "telemwait", 500, 5000, 1000);
else
interval = 1000;
break;
case DLY_ID:
if(wait_times)
interval = retrieve_astcfgint(wait_times_save, "idwait",250,5000,500);
else
interval = 500;
break;
case DLY_UNKEY:
if(wait_times)
interval = retrieve_astcfgint(wait_times_save, "unkeywait",500,5000,1000);
else
interval = 1000;
break;
case DLY_CALLTERM:
if(wait_times)
interval = retrieve_astcfgint(wait_times_save, "calltermwait",500,5000,1500);
else
interval = 1500;
break;
default:
return 0;
}
int interval;
char *wait_times;
char *wait_times_save = NULL;
wait_times = ast_variable_retrieve(cfg, myrpt->name, "wait_times");
if (wait_times) {
wait_times_save = ast_strdupa(wait_times);
if (!wait_times_save) {
ast_log(LOG_WARNING, "Out of memory in wait_interval()\n");
wait_times = NULL;
}
}
switch (type) {
case DLY_TELEM:
if (wait_times)
interval = retrieve_astcfgint(wait_times_save, "telemwait", 500, 5000, 1000);
else
interval = 1000;
break;
case DLY_ID:
if (wait_times)
interval = retrieve_astcfgint(wait_times_save, "idwait", 250, 5000, 500);
else
interval = 500;
break;
case DLY_UNKEY:
if (wait_times)
interval = retrieve_astcfgint(wait_times_save, "unkeywait", 500, 5000, 1000);
else
interval = 1000;
break;
case DLY_CALLTERM:
if (wait_times)
interval = retrieve_astcfgint(wait_times_save, "calltermwait", 500, 5000, 1500);
else
interval = 1500;
break;
default:
return 0;
}
return interval;
}
}
/*
@@ -1522,6 +1521,7 @@ pthread_attr_t attr;
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
ast_pthread_create(&tele->threadid,&attr,rpt_tele_thread,(void *) tele);
pthread_attr_destroy(&attr);
return;
}
@@ -2154,6 +2154,7 @@ static int function_autopatchup(struct rpt *myrpt, char *param, char *digitbuf,
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
ast_pthread_create(&myrpt->rpt_call_thread,&attr,rpt_call,(void *) myrpt);
pthread_attr_destroy(&attr);
return DC_COMPLETE;
}
@@ -5240,6 +5241,7 @@ char cmd[MAXDTMF+1] = "";
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
ast_pthread_create(&myrpt->rpt_call_thread,&attr,rpt_call,(void *)myrpt);
pthread_attr_destroy(&attr);
continue;
}
}
@@ -5782,6 +5784,7 @@ pthread_attr_t attr;
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
ast_pthread_create(&rpt_vars[i].rpt_thread,&attr,rpt,(void *) &rpt_vars[i]);
pthread_attr_destroy(&attr);
}
usleep(500000);
for(;;)
@@ -5817,6 +5820,7 @@ pthread_attr_t attr;
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
ast_pthread_create(&rpt_vars[i].rpt_thread,&attr,rpt,(void *) &rpt_vars[i]);
pthread_attr_destroy(&attr);
ast_log(LOG_WARNING, "rpt_thread restarted on node %s\n", rpt_vars[i].name);
}

0
apps/app_sayunixtime.c Executable file → Normal file
View File

2
apps/app_senddtmf.c Executable file → Normal file
View File

@@ -50,7 +50,7 @@ static char *synopsis = "Sends arbitrary DTMF digits";
static char *descrip =
" SendDTMF(digits[|timeout_ms]): Sends DTMF digits on a channel. \n"
" Accepted digits: 0-9, *#abcd\n"
" Accepted digits: 0-9, *#abcd, w (.5s pause)\n"
" The application will either pass the assigned digits or terminate if it\n"
" encounters an error.\n";

0
apps/app_sendtext.c Executable file → Normal file
View File

0
apps/app_setcallerid.c Executable file → Normal file
View File

0
apps/app_setcdruserfield.c Executable file → Normal file
View File

0
apps/app_setcidname.c Executable file → Normal file
View File

0
apps/app_setcidnum.c Executable file → Normal file
View File

0
apps/app_setrdnis.c Executable file → Normal file
View File

0
apps/app_settransfercapability.c Executable file → Normal file
View File

9
apps/app_skel.c Executable file → Normal file
View File

@@ -3,7 +3,7 @@
*
* Copyright (C) <Year>, <Your Name Here>
*
* <Your Name Here> <<You Email Here>>
* <Your Name Here> <<Your Email Here>>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
@@ -20,7 +20,7 @@
*
* \brief Skeleton application
*
* This is a skeleton for development of an Asterisk application */
* This is a skeleton for development of an Asterisk application
* \ingroup applications
*/
@@ -53,7 +53,7 @@ static char *descrip = "This application is a template to build other applicatio
#define OPTION_C (1 << 2) /* Option C(str) */
#define OPTION_NULL (1 << 3) /* Dummy Termination */
AST_DECLARE_OPTIONS(app_opts,{
AST_APP_OPTIONS(app_opts,{
['a'] = { OPTION_A },
['b'] = { OPTION_B, 1 },
['c'] = { OPTION_C, 2 }
@@ -77,7 +77,6 @@ static int app_exec(struct ast_channel *chan, void *data)
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "%s requires an argument (dummy|[options])\n",app);
LOCAL_USER_REMOVE(u);
return -1;
}
@@ -96,7 +95,7 @@ static int app_exec(struct ast_channel *chan, void *data)
if ((argc = ast_app_separate_args(args, '|', argv, sizeof(argv) / sizeof(argv[0])))) {
dummy = argv[0];
options = argv[1];
ast_parseoptions(app_opts, &flags, opts, options);
ast_app_parse_options(app_opts, &flags, opts, options);
}
if (!ast_strlen_zero(dummy))

33
apps/app_sms.c Executable file → Normal file
View File

@@ -694,7 +694,7 @@ static void sms_readfile (sms_t * h, char *fn)
}
while (fgets (line, sizeof (line), s))
{ /* process line in file */
char *p;
unsigned char *p;
for (p = line; *p && *p != '\n' && *p != '\r'; p++);
*p = 0; /* strip eoln */
p = line;
@@ -1050,7 +1050,7 @@ static void sms_nextoutgoing (sms_t * h)
unsigned char p = 2;
h->omsg[0] = 0x91; /* SMS_DATA */
if (h->smsc) { /* deliver */
h->omsg[p++] = (more ? 4 : 0);
h->omsg[p++] = (more ? 4 : 0) + ((h->udhl > 0) ? 0x40 : 0);
p += packaddress (h->omsg + p, h->oa);
h->omsg[p++] = h->pid;
h->omsg[p++] = h->dcs;
@@ -1178,25 +1178,24 @@ static void sms_messagetx(sms_t * h)
static int sms_generate (struct ast_channel *chan, void *data, int len, int samples)
{
struct ast_frame f = { 0 };
unsigned char waste[AST_FRIENDLY_OFFSET];
#define MAXSAMPLES 800
#ifdef OUTALAW
unsigned char buf[800];
unsigned char *buf;
#else
signed short buf[800];
short *buf;
#endif
#define SAMPLE2LEN sizeof(*buf)
sms_t *h = data;
int i;
if (len > sizeof (buf)) {
ast_log (LOG_WARNING, "Only doing %d bytes (%d bytes requested)\n", (int)(sizeof (buf) / sizeof (signed short)), len);
len = sizeof (buf);
#ifdef OUTALAW
samples = len;
#else
samples = len / 2;
#endif
if (samples > MAXSAMPLES) {
ast_log (LOG_WARNING, "Only doing %d samples (%d requested)\n",
MAXSAMPLES, samples);
samples = MAXSAMPLES;
}
waste[0] = 0; /* make compiler happy */
len = samples * SAMPLE2LEN + AST_FRIENDLY_OFFSET;
buf = alloca(len);
f.frametype = AST_FRAME_VOICE;
#ifdef OUTALAW
f.subclass = AST_FORMAT_ALAW;
@@ -1379,8 +1378,8 @@ static int sms_exec (struct ast_channel *chan, void *data)
ast_copy_string (h.cli, chan->cid.cid_num, sizeof (h.cli));
{
char *d = data,
*p,
unsigned char *p;
unsigned char *d = data,
answer = 0;
if (!*d || *d == '|') {
ast_log (LOG_ERROR, "Requires queue name\n");
@@ -1449,7 +1448,7 @@ static int sms_exec (struct ast_channel *chan, void *data)
d = p;
h.udl = 0;
while (*p && h.udl < SMSLEN)
h.ud[h.udl++] = utf8decode((unsigned char **)&p);
h.ud[h.udl++] = utf8decode(&p);
if (is7bit (h.dcs) && packsms7 (0, h.udhl, h.udh, h.udl, h.ud) < 0)
ast_log (LOG_WARNING, "Invalid 7 bit GSM data\n");
if (is8bit (h.dcs) && packsms8 (0, h.udhl, h.udh, h.udl, h.ud) < 0)

0
apps/app_softhangup.c Executable file → Normal file
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Some files were not shown because too many files have changed in this diff Show More