- 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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48200 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48196 f38db490-d61c-443f-a65b-d21fe96a405b
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48194 f38db490-d61c-443f-a65b-d21fe96a405b
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48191 f38db490-d61c-443f-a65b-d21fe96a405b
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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48148 f38db490-d61c-443f-a65b-d21fe96a405b
- 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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48110 f38db490-d61c-443f-a65b-d21fe96a405b
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48109 f38db490-d61c-443f-a65b-d21fe96a405b
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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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48090 f38db490-d61c-443f-a65b-d21fe96a405b
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 ?
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48084 f38db490-d61c-443f-a65b-d21fe96a405b
to simplify the body of the main loop of the accepting thread.
Rename purge_unused() to purge_events() so one knows what the
function does.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48081 f38db490-d61c-443f-a65b-d21fe96a405b
+ use a wrapper around ast_carefulwrite(), used in two places,
to make life easier when we decide to use a different interface
to the socket.
+ put an ast_verbose() message on astman_append on a case that
should never happen now that we use a temporary file for
AMI-over-HTTP sessions
+ document and slightly simplify process_events() by removing
unnecessary parentheses.
+ in get_input(), use ast_wait_for_input() instead of poll().
We may want to move to a completely non-blocking
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48080 f38db490-d61c-443f-a65b-d21fe96a405b
causing the http threads to do busy waiting around the socket...
Fix the mistake, sorry for the inconvenience!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48074 f38db490-d61c-443f-a65b-d21fe96a405b
opening an SSL socket.
We are almost ready to make this code available to other modules.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48073 f38db490-d61c-443f-a65b-d21fe96a405b
oej