mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
64 Commits (a858c92e3f973b5a4abe44e42683c8b87b51cc4d)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Steve Murphy
|
09c0d56c5c |
Merged revisions 59522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59522 | murf | 2007-03-30 11:51:17 -0600 (Fri, 30 Mar 2007) | 1 line several changes via kpflemings review ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Steve Murphy
|
0f11d3c8c3 |
Merged revisions 59486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59486 | murf | 2007-03-30 08:11:59 -0600 (Fri, 30 Mar 2007) | 1 line These mods fix CDR issues from 8221, 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated from transfer situations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Tilghman Lesher
|
590cb3a6fa |
Expand datastores to add the notion of inheritance. This will be needed for
the conversion of IAX2 variables from the current custom method to ast_storage. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Russell Bryant
|
b4a29c3782 |
Constify the list of codec preferences.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57293 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Tilghman Lesher
|
b0f60e7496 |
Merged revisions 56685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56685 | tilghman | 2007-02-25 08:46:41 -0600 (Sun, 25 Feb 2007) | 11 lines Merged revisions 56684 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) | 3 lines Issue 9130 - If prev is the last item on the channel list, then evaluating additional conditions (e.g. name prefix) will cause a NULL dereference. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56686 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Olle Johansson
|
75d387acbc |
Doxygen additions, corrections
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Joshua Colp
|
afc99294fa |
Merged revisions 56231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, 22 Feb 2007) | 10 lines Merged revisions 56230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56232 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
ba32ee49d0 |
Adding Realtime Text support (T.140) to Asterisk
T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
8470ee5cd0 |
Merged revisions 54290 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54290 | file | 2007-02-13 20:09:40 -0500 (Tue, 13 Feb 2007) | 2 lines Add G722 to ast_best_codec. If anyone disagrees with it's placement, feel free to change it. (issue #9045 reported by gork) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54291 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
10c4a5fef7 |
Simplify a small bit of logic.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Paul Cadach
|
85ad583544 |
Merged revisions 53879 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53879 | pcadach | 2007-02-10 01:07:11 -0800 (Сбт, 10 Фев 2007) | 1 line Provide correct DTMF duration ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
dab41a355d |
Merged revisions 51848 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r51848 | russell | 2007-01-23 18:59:58 -0600 (Tue, 23 Jan 2007) | 14 lines Merged revisions 51843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r51843 | russell | 2007-01-23 18:57:28 -0600 (Tue, 23 Jan 2007) | 6 lines Fix an issue related to synchronization of recordings when using Monitor(). The bug is a miscalculation of the amount to seek the stream for writing to disk when the number of samples coming in and out of a channel do not match up. (issue #8298, #8887, report and patch by guillecabeza, patch files created and testing done by whoiswes) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51850 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
21b53af31d |
Cosmetic changes. Make main source files better conform to coding guidelines and standards. (issue #8679 reported by johann8384)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51486 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
dcca8f345f |
Merged revisions 51311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Luigi Rizzo
|
5b9114fa73 |
include "asterisk/zapata.h" to get the zaptel headers.
this should be the last one left around... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51299 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Jason Parker
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11dd11e5a1 |
Merged revisions 51241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51241 | qwell | 2007-01-18 12:28:29 -0600 (Thu, 18 Jan 2007) | 2 lines Fix an issue with deprecated commands ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51242 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
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c71d6c12f7 |
Don't hold channel lock while sleeping/waiting for audio stream to get setup. (issue #8834 reported by phsultan)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51193 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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d986f00e73 |
Merged revisions 50727 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50727 | file | 2007-01-13 01:00:24 -0500 (Sat, 13 Jan 2007) | 2 lines Only write a frame out to the channel if one exists. There are cases where one may not and would therefore cause the channel driver to segfault. (issue #8434 reported by slimey) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50728 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Kevin P. Fleming
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17ea9c930e |
make the automatic post-answer delay happen only when the answer is 'automatic' (not done by the Answer() dialplan application)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50571 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Kevin P. Fleming
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1439e0fa75 |
when a channel gets automatically answered by an application, sleep a bit to give the audio path (for VOIP channels) time to be setup
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Tilghman Lesher
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33d5a8a582 |
Reduce duplication of code (Issue 6542)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49784 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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a9c3429b07 |
Merged revisions 49675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49675 | file | 2007-01-05 17:14:47 -0500 (Fri, 05 Jan 2007) | 2 lines Don't keep repeating the warning over and over when the end of the call is reached. (issue #8724 reported by xrg) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Kevin P. Fleming
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3f7899c9da |
small formatting fix
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49068 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Kevin P. Fleming
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adca0ff14b |
Merged revisions 49006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49006 | kpfleming | 2006-12-27 16:06:56 -0600 (Wed, 27 Dec 2006) | 2 lines since these variables all have static duration, none of them need initializers (they default to zero anyway) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49008 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Luigi Rizzo
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09f75aa6dc |
rename the structs struct tone_zone_sound and struct tone_zone
defined in indications.h to ind_tone_zone_sound and ind_tone_zone, to avoid conflicts with the structs with the same names defined in tonezone.h Hope i haven't missed any instance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48958 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Luigi Rizzo
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a2e6c6277f |
same as in other places, check that generator->release is not NULL
before calling it. This allows generators to set it to NULL when they have nothing to do there. Later, the three copies of the code that releases a generator should be moved to a function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48766 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Luigi Rizzo
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b6d1722c83 |
remove ast_safe_string_alloc() - it is completely
equivalent to asprintf(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48499 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Luigi Rizzo
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1122621981 |
constify ast_state2str() and note it is not reentrant.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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17a2888d2e |
Staticize one, and Constify a bunch of usage strings for CLI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48303 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Olle Johansson
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e3b099c12a |
Formatting fix
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Luigi Rizzo
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d7b26b6bf8 |
set pointers to NULL after freeing memory to avoid multiple free()
probably 1.4/1.2 issue as well if someone can look into that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48001 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Steve Murphy
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6dcb17baaf |
This update fulfils the request of bug 7109, which claimed the language arg to ast_stream_and_wait() was redundant. Almost all calls just used chan->language, and seeing how chan is the first argument, this certainly seems redundant. A change of language could just as easily be done by simply changing the channel language before calling.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Paul Cadach
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fc58bec502 |
Merged revisions 44809 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44809 | pcadach | 2006-10-10 23:44:54 +0700 (Втр, 10 Окт 2006) | 1 line CHANNEL() function sometime mix parameter and value ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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0f15e43add |
Merged revisions 47707 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47707 | file | 2006-11-15 16:33:41 -0500 (Wed, 15 Nov 2006) | 2 lines We need to ensure timelimit stuff is included as well so warnings get played. (issue #8050 reported by KNK) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Steve Murphy
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2e375b388e |
This mod via bug 7531
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Steve Murphy
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908f176cf3 |
A fair number of changes for the sake of bug 7506
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47290 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Tilghman Lesher
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10875731ec |
Merged revisions 47051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006) | 2 lines Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments" ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47052 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Tilghman Lesher
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a8d3ee89c6 |
Merged revisions 46078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46078 | tilghman | 2006-10-23 22:01:00 -0500 (Mon, 23 Oct 2006) | 3 lines Pass through a frame if we don't know what it is, rather than trying to pass a NULL, which will segfault a channel driver (Bug 8149) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46079 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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bd53e7ee4c |
Extend the thread storage API such that a custom initialization function can
be called for each thread specific object after they are allocated. Note that there was already the ability to define a custom cleanup function. Also, if the custom cleanup function is used, it *MUST* call free on the thread specific object at the end. There is no way to have this magically done that I can think of because the cleanup function registered with the pthread implementation will only call the function back with a pointer to the thread specific object, not the parent ast_threadstorage object. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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40b8afd54f |
Merged revisions 45441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45441 | russell | 2006-10-17 22:41:36 -0400 (Tue, 17 Oct 2006) | 7 lines Don't attempt to access private data members of the pthread_mutex_t object, because this does not work on all linux systems. Instead, just access the reentrancy field in the ast_mutex_info struct when DEBUG_THREADS is enabled. If DEBUG_CHANNEL_LOCKS is enabled, the developer probably has DEBUG_THREADS on as well. (issue #8139, me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45442 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Kevin P. Fleming
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696f9ed677 |
Merged revisions 45408 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45408 | kpfleming | 2006-10-17 17:24:10 -0500 (Tue, 17 Oct 2006) | 3 lines optimize the 'quick response' code a bit more... no more malloc() or memset() for each response expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45409 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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b5f2589e33 |
Add Masquerade manager event which trips when a masquerade happens (issue #7840 reported by moy)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44273 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Matt O'Gorman
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ae8cc3e18b |
bug #8076 check option_debug before printing to debug channel.
patch provided in bugnote, with minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Joshua Colp
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3e4a081e1c |
Make callerid fields in Manager events more consistent. CallerIDNum for number and CallerIDName for name. (issue #7976 reported by suhler)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44217 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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Russell Bryant
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d2c57c5f4f |
Merged revisions 43779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r43779 | russell | 2006-09-27 12:55:49 -0400 (Wed, 27 Sep 2006) | 50 lines Merged revisions 43778 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines Fix a problem that occurred if a user entered a digit that matched a bridge feature that was configured using multiple digits, and the digit that was pressed timed out in the feature digit timeout period. For example, if blind transfer is configured as '##', and a user presses just '#'. In this situation, the call would lock up and no longer pass any frames. (issue #7977 reported by festr, and issue #7982 reported by michaels and valuable input provided by mneuhauser and kuj. Fixed by me, with testing help and peer review from Joshua Colp). There are a couple of issues involved in this fix: 1) When ast_generic_bridge determines that there has been a timeout, it returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls ast_generic_bridge over again with the same timestamp for the next event. This results in an endless loop of nothing until the call is terminated. This is resolved by simply changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it sees a timeout. 2) I also changed ast_channel_bridge such that if in the process of calculating the time until the next event, it knows a timeout has already occured, to immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the channels anyway. 3) In the process of testing the previous two changes, I ran into a problem in res_features where ast_channel_bridge would return because it determined that there was a timeout. However, ast_bridge_call in res_features would then determine by its own calculation that there was still 1 ms before the timeout really occurs. It would then proceed, and since the bridge broke out and did *not* return a frame, it interpreted this as the call was over and hung up the channels. The reason for this was because ast_bridge_call in res_features and ast_channel_bridge in channel.c were using different times for their calculations. channel.c uses the start_time on the bridge config, which is the time that the feature digit was recieved. However, res_features had another time, 'start', which was set right before calling ast_channel_bridge. 'start' will always be slightly after start_time in the bridge config, and sometimes enough to round up to one ms. This is fixed by making ast_bridge_call use the same time as ast_channel_bridge for the timeout calculation. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43780 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
|
Joshua Colp
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becd11ddb8 |
Merged revisions 43695 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2 lines Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43696 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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Kevin P. Fleming
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acf824fe5e |
Merged revisions 43486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43486 | kpfleming | 2006-09-22 10:51:13 -0500 (Fri, 22 Sep 2006) | 2 lines all the Linux systems I have don't use '__m_count' for this field, so I don't know where this came from... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43488 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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Tilghman Lesher
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2b55678e1f |
Remove deprecated CLI apps from the core
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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Joshua Colp
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1c764935f2 |
SS7 marked the start of an open season for trunk again but here's something minor - abstract early bridging into the technology so that we don't always assume they use RTP and try it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43437 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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Kevin P. Fleming
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fcb999c01c |
merge qwell's CLI verbification work
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |