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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48223 | russell | 2006-12-03 01:34:14 -0500 (Sun, 03 Dec 2006) | 3 lines
When "fetch" is in use, instead of "wget", --continue is not a valid option.
(issue #8451)
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make depend again (or ?)
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This is imported from branch "invitestate" and "invitestate-1.4"
***
***
*** IF YOU HAVE ISSUES WITH BYEs/CANCELs - PLEASE UPDATE AND TEST AGAIN!
*** Thank you!
***
***
/Olle
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Something is wrong in the agi directory. Asking for autoconfig.h.
I have it disabled locally, but no reason to commit that change.
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- remove coef_in.h and coef_out.h that was only included as data definitions in fskmodem.c
If you understand spanish, please help us translate the comments in fskmodem.c. Thanks!
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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r48195 | russell | 2006-12-01 22:50:58 -0500 (Fri, 01 Dec 2006) | 3 lines
Backport the comment containing the warning regarding the limitations on the
usage of this function. It is thread safe, but not technically reentrant.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48193 | kpfleming | 2006-12-01 17:37:28 -0600 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines
if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48190 | russell | 2006-12-01 18:16:28 -0500 (Fri, 01 Dec 2006) | 12 lines
FreeBSD 6.1 does not include wget by default. However, it has fetch which will
work just fine for our purposes of downloading the sounds packages. So, check
for both wget and fetch and the configure script and use what was found to
download them. If neither one was found, and sound packages are selected that
must be downloaded, the install process will print out an informative error
message indicating the situation.
Also, fix a couple places where "make" was hard coded into some output messages
by replacing them with the $(MAKE) variable.
(issue #8451, initial patch by pabelanger, with additional modifications by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48186 | qwell | 2006-12-01 14:25:51 -0600 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48183 via svnmerge from
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r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines
Fix a small typo - issue 8848, reported by pabelanger
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48179 | tilghman | 2006-12-01 13:38:59 -0600 (Fri, 01 Dec 2006) | 2 lines
Double-unlock error (reported by blitzrage on IRC)
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(imported from 1.4)
- Small fixes to limitonpeer
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)
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r48162 | file | 2006-11-30 15:28:19 -0500 (Thu, 30 Nov 2006) | 9 lines
Blocked revisions 48161 via svnmerge
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r48161 | file | 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines
Don't write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel driver. (issue #8390 reported by hselasky)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48158 | file | 2006-11-30 15:07:55 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48157 via svnmerge from
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r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 lines
Only print out debug message if bridged channel is not NULL. (issue #8412 reported by jubilex)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48155 | file | 2006-11-30 14:05:14 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48154 via svnmerge from
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r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 lines
Do not listen for DTMF on the bridge that comes into existence when ParkedCall is executed. This means native bridging can now occur for this. (issue #8406 reported by kebl0155)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48152 | file | 2006-11-30 13:47:40 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48151 via svnmerge from
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r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 lines
Print certain CDR messages out at the NOTICE level versus WARNING since they can occur when used with the CDR applications and are perfectly fine. (issue #8367 reported by dartvader)
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r48147 | file | 2006-11-30 13:19:55 -0500 (Thu, 30 Nov 2006) | 9 lines
Blocked revisions 48146 via svnmerge
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r48146 | file | 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines
Remember the pointer to the allocated block of memory so that we can free it and not cause a memory leak. (issue #8449 reported by arkadia)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48142 via svnmerge from
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r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines
Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage)
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Luigi, but hit me in the back today.
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source, not the configuration option name so let's turn it back to srvlookup instead of global_srvlookup. (issue #8442 reported by jtodd)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48115 | file | 2006-11-29 16:05:17 -0500 (Wed, 29 Nov 2006) | 2 lines
Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420 reported by slimey)
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- Add some more documentation for the ast_dynamic_str_............() function
to document the behavior of the function in the case of a partial write.
Also, document the return value and note that the function should never need
to be called directly.
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in both the append and non-append modes. Also, always truncating the partial
write makes the behavior of the function more consistent, where in any type of
write, no partial result is left in the buffer.
Thanks for the feedback, luigi
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48107 | file | 2006-11-29 11:50:33 -0500 (Wed, 29 Nov 2006) | 10 lines
Merged revisions 48106 via svnmerge from
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r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines
If the frame was duplicated before writing out then we need to free it. (issue #8429 reported by edguy3)
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and also add a more verbose comment explaining why it is only needed in the
case of appending to the string for any curious readers that come along in the
future.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48101 | file | 2006-11-28 23:26:53 -0500 (Tue, 28 Nov 2006) | 2 lines
Don't crash if the mailstream was not created.
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that occurs for every incoming media frame.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48095 | qwell | 2006-11-28 12:26:53 -0600 (Tue, 28 Nov 2006) | 2 lines
Export several more variables in top level Makefile. Inspired by issue 8438.
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an extra allocation on a path where we have way too many already.
Unfortunately the AMI-over-HTTP requires multiple copies,
because we need to generate a header, then the raw output to
an intermediate buffer, then convert it to html/xml, and
finally copy everything into a malloc'ed buffer because
that's what the generic_http_callback interface expects.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48088 | file | 2006-11-28 11:57:16 -0500 (Tue, 28 Nov 2006) | 10 lines
Merged revisions 48087 via svnmerge from
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r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines
According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3)
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ast_dynamic_str_thread_build_va_couldnt_we_choose_a_shorter_name()
I am unsure whether the truncation of the string in case of a failed
attempt should be done unconditionally. See the XXX mark.
Russel, ideas ?
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no output.
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