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r56505 | russell | 2007-02-23 17:24:18 -0600 (Fri, 23 Feb 2007) | 16 lines
Merged revisions 56504 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56504 | russell | 2007-02-23 17:20:55 -0600 (Fri, 23 Feb 2007) | 8 lines
Fix up a couple more signal handlers to not do bad things that could cause
various undesirable results. The other day, I made Asterisk deadlock by
hitting Control-C because of a bad signal handler. Now, signal handlers
just set a flag and write to an alert pipe for the flag to be handled. Then,
there is another thread that is monitoring for these flags. If being run in
console mode, it is just the main thread. If Asterisk is in the background,
a thread is created to do it.
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r56457 | file | 2007-02-23 16:53:41 -0500 (Fri, 23 Feb 2007) | 2 lines
Change log notice to debug. It is possible for a scheduled item to execute and be deleted at close to the same time and unavoidable. If this happens this message creeps up.
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performance of the GUI. This encodes the configuration into the JSON format
in a manager header, "JSON: ". The encoded information can be directly used
as a javascript object, so no parsing is needed. For large configuration
files, this can greatly improve loading times in the GUI. Furthermore, the
encoding takes up a lot less space when being transmitted than the other
alternatives. (Inspired by discussion with Pari)
Here is an example of what you get:
http://localhost:8088/asterisk/rawman?action=getconfigjson&filename=users.conf
Response: Success
JSON: {"general":["hasvoicemail=yes"],"6000":["fullname=russell","secret=1234"]}
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r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) | 18 lines
Merge changes from team/russell/sla_updates.
This batch of changes to the SLA code does a few different things.
* I made the SLA code event driven instead of having to act in a lot of busy
loops while dialing things to wait for state changes. This makes the code
more efficient and readable at the same time.
* I have implemented a couple of new features. The first is inbound trunk
ringing timeouts. This is an option that defines how long to let an incoming
call on a trunk to ring.
* I have also implemented ring timeouts for stations. They may be specified
for the entire station, meaning it is how long to let the station ring before
giving up. You can also specify a ring timeout for a specific trunk on a
station. So, you can say that you only want a specific station to ring 5
seconds if it is line1 ringing, but otherwise, there is no timeout.
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convert various #if expressions to #ifdef for macros that may not be defined (and where the value is not important)
Note: two of these changes are in bison generated files which is going to be inconvenient when they are regenerated
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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.
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pretty cool things.
First, you can get the device state of anything in the dialplan:
NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)})
NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)})
Most importantly, this allows you to create custom device states so you can
control phone lamps directly from the dialplan.
Set(DEVSTATE(Custom:mycustomlamp)=BUSY)
...
exten => mycustomlamp,hint,Custom:mycustomlamp
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r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines
- Add the ability to register a callback to monitor state changes in an
asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API
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r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines
Merge team/russell/sla_rewrite
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
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r53497 | russell | 2007-02-07 17:52:45 -0600 (Wed, 07 Feb 2007) | 6 lines
When building libdb1.a, put the additional flags needed at the beginning of
ASTCFLAGS, instead of at the end. This way, we ensure that we find the local
headers first before accidentally trying to use headers that exist in
locations specified in the ASTCFLAGS passed from the main Makefile.
(issue #8637, ovi)
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r53464 | russell | 2007-02-07 14:07:39 -0600 (Wed, 07 Feb 2007) | 4 lines
The clean target actually needs to run "distclean" on editline. This is
because we need to make sure that its configure script gets executed again,
because the CFLAGS we want to pass to editline may have changed.
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r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) | 7 lines
When parsing the NTP timestamp in a sender report message, you are supposed to
take the low 16 bits of the integer part, and the high 16 bits of the
fractional part. However, the code here was erroneously taking the low 16 bits
of the fractional part. It then shifted the result 16 bits down, so the result
was always zero. This fix makes it grab the appropriate high 16 bits, instead.
(issue #8991, pointed out by andre_abrantes)
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r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 lines
When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party.
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r52904 | russell | 2007-01-30 11:19:39 -0600 (Tue, 30 Jan 2007) | 17 lines
Merged revisions 52903 via svnmerge from
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r52903 | russell | 2007-01-30 11:12:04 -0600 (Tue, 30 Jan 2007) | 9 lines
The SIGHUP handler was implemented to allow admins to send SIGHUP to a running
Asterisk process to reload the configuration. However, doing the actual reload
in the signal handler itself is a very bad thing to do, because the reload
process includes calling non-reentrant functions such as malloc/calloc/etc.
If Asterisk is running in the background, then the reload will happen
immediately. However, if running in console mode, the reload doesn't work
until something is typed at the console. That sort of defeats the purpose,
but I don't see an easy way to get around it at this point.
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r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) | 6 lines
Fix a problem with packet-to-packet bridging and DTMF mode translation. P2P
bridging can only be used when the DTMF modes don't match if the core is
monitoring DTMF in both directions. Then, the core will handle the translation.
Otherwise, this bridging method can not be used.
(issue #8936)
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- changing the actionlock to a rwlock
- not locking the session before doing the action callback
The crash issue in 8711 should not be an issue here.
Merged revisions 52611 via svnmerge from
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r52611 | russell | 2007-01-29 14:39:20 -0600 (Mon, 29 Jan 2007) | 10 lines
The session lock can not be held while calling action callbacks. If so, then
when the WaitEvent callback gets called, then no event can happen because the
session can't be locked by another thread. Also, the session needs to be
locked in the HTTP callback when it reads out the output string. This fixes
the deadlock reported in both 8711 and 8934.
Regarding issue 8711, there still may be an issue. If there is a second action
requested before the processing of the first action is finished, there could
still be some corruption of the output string buffer used to build the result.
(issue #8711, #8934)
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r52494 | jdixon | 2007-01-28 22:18:36 -0600 (Sun, 28 Jan 2007) | 4 lines
Fixed problem with jitterbuf, whereas it would not complain about, and
would allow itself to be overfilled (per the max_jitterbuf parameter). Now
it rejects any data over and above that size, and complains about it.
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r52506 | russell | 2007-01-29 10:54:27 -0600 (Mon, 29 Jan 2007) | 5 lines
Clean up a few things in the last commit to the adaptive jitterbuffer code.
- Specifically indicate to the compiler that the "dropem" variable only
needs one but.
- Change formatting to conform to coding guidelines.
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r51848 | russell | 2007-01-23 18:59:58 -0600 (Tue, 23 Jan 2007) | 14 lines
Merged revisions 51843 via svnmerge from
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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)
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r51781 | russell | 2007-01-23 16:04:01 -0600 (Tue, 23 Jan 2007) | 6 lines
Fix some bugs in process_message(). The manager session lock needs to be held
when sending some sort of response, or calling one of the manager action
callbacks. This resolves an issue where people using the GUI would get random
crashes when they start clicking around a lot.
(issue #8711, reported and debugged by zandbelt)
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r51750 | russell | 2007-01-23 15:33:15 -0600 (Tue, 23 Jan 2007) | 4 lines
When traversing the list of manager actions, the iterator needs to be
initialized to the list head *after* locking the list. Also, lock the actions
list in one place it is being accessed where it was not being done.
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r51683 | murf | 2007-01-23 11:58:27 -0700 (Tue, 23 Jan 2007) | 1 line
via 8748 (callerid.c loses name when returning PRIVATE_NUMBER flag), the user suggested this mod, saying it would allow 'WITHHELD' to appear in the name field, which would be useful
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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...)
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has been found so that 'cat' is non-NULL. This way, users.conf is only checked
when necessary. (issue #8852, akohlsmith, committed patch a bit different)
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r51262 | russell | 2007-01-18 15:54:23 -0600 (Thu, 18 Jan 2007) | 5 lines
Ensure that the locations given to the Asterisk configure script for ncurses,
curses, termcap, or tinfo are further passed along to the editline configure
script. This fixes some cross-compilation environments.
(issue #8637, reported by ovi, patch by me)
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r51195 | tilghman | 2007-01-17 14:56:15 -0600 (Wed, 17 Jan 2007) | 12 lines
Merged revisions 51194 via svnmerge from
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r51194 | tilghman | 2007-01-17 14:52:21 -0600 (Wed, 17 Jan 2007) | 4 lines
When ast_strip_quoted was called with a zero-length string, it would treat a
NULL as if it were the quoting character (and would thus return the string
in memory immediately following the passed-in string).
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r50867 | kpfleming | 2007-01-15 09:03:06 -0600 (Mon, 15 Jan 2007) | 2 lines
use the ACX_PTHREAD macro from the Autoconf macro archive for setting up compiler pthreads support... should improve portability to platforms with unusual pthreads requirements
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r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines
Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson)
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previously set are erroneously still set (Bug 6701). After discussion,
it was determined this should only be changed in trunk.
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r49742 | qwell | 2007-01-05 18:24:38 -0600 (Fri, 05 Jan 2007) | 7 lines
Save 1 whopping byte of allocated memory!
This looks like it may have been a chicken/egg scenario..
You had to call a cleanup func, because everything was allocated.
Then since you had to call a cleanup func, you were forced to allocate - ie; strdup("").
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r49457 | kpfleming | 2007-01-04 12:05:47 -0600 (Thu, 04 Jan 2007) | 2 lines
make building of codec_gsm against the system GSM library actually work
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r49460 | kpfleming | 2007-01-04 12:16:40 -0600 (Thu, 04 Jan 2007) | 2 lines
don't define this type either if LOW_MEMORY is enabled
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r49461 | kpfleming | 2007-01-04 12:17:01 -0600 (Thu, 04 Jan 2007) | 2 lines
don't do frame header caching in the core if LOW_MEMORY is defined
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r49066 | file | 2006-12-30 00:46:57 -0500 (Sat, 30 Dec 2006) | 2 lines
If the Packet2Packet bridge is being broken because of a masquerade then attempt to read a frame in so the masquerade actually happens. Otherwise weirdness will occur. (issue #8696 reported by kjotte)
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r49009 | file | 2006-12-27 17:28:46 -0500 (Wed, 27 Dec 2006) | 2 lines
ast_copy_string is not available when LOW_MEMORY is used and things are being built in the utils directory, so we need to resort to the old method of strncpy. (issue #8579 reported by mottano)
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r48998 | kpfleming | 2006-12-27 15:08:30 -0600 (Wed, 27 Dec 2006) | 3 lines
move extern declaration for this option to a header file where it belongs
provide an initial value for 'languageprefix' option, instead of relying on randomness to provide a useful value
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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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.
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done for exactly how much memory is needed. This was suggested by Luigi on
the asterisk-dev mailing list. Thanks!
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this, implementing locking of this list to make it thread-safe.
- Add a "redirect" option to http.conf that allows redirecting one URI to
another. I was inspired to do this while playing with the Asterisk GUI. I
got tired of typing this URL to get to the GUI:
http://localhost:8088/asterisk/static/config/cfgadvanced.html
So, now I have the following line in http.conf:
redirect=/=/asterisk/static/config/cfgadvanced.html
Now, I can type the following into my browser and go to the GUI:
http://localhost:8088
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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.
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On passing remove presumably duplicate code to generate
the message for the manager_hooks:
in the previous version, the message was almost the same as the one sent
to regular sessions, with the exception of the empty line at the end, and
a few (presumably unintentional) differences e.g. timestamps,
debugging, and lowercase headers for "event" and "privilege".
now we reuse the same message as before.
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r48525 | kpfleming | 2006-12-16 15:14:34 -0600 (Sat, 16 Dec 2006) | 2 lines
simplify dependency tracking system, using the compiler's built-in method for generating them, and only doing dependency tracking if developer mode is enabled via the configure script
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r48521 | kpfleming | 2006-12-16 14:12:41 -0600 (Sat, 16 Dec 2006) | 2 lines
since we really, really have to have autoconfig.h included before all other headers (especially system headers), the Makefile will now force it to happen (this will fix build problems with files like ast_expr2f.c, where we can't control the inclusion order in the file itself)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
renaming them to ast_str ... and putting the
struct ast_threadstorage pointer into the struct ast_str.
This makes the code a lot more readable.
At this point we can use these routines also to
replace ast_build_string().
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While doing this, add a bit of documentation, and slightly
extend the functionality as follows:
+ a max_len of -1 means that we take whatever the current size
is, and never try to extend the buffer;
+ add support for alloca()-ted dynamic strings, which is very
useful for all cases where we do an ast_build_string() now.
Next step is to simplify the interface by using shorter names
(e.g. ast_str as a prefix) and removing the _thread variant
of the functions by saving the threadstorage reference into
the struct ast_str. This can be done by overloading the
'type' field.
Finally, I will do my best to remove the convoluted interface
that results from trying to support platforms without va_copy().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Not long ago i replaced lseek() with fseek() but
forgot that filr FILE's you need ftell to
give you the current position.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2 lines
Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman)
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reducing indentation and normalizing loops.
While doing this, remove some unused variables,
fix an uninitialized string (idaction), and mark
some places where the behaviour is not what we would expect
(e.g. an empty context is reported as an error same as
a non-existent one). Given that this function is
not in 1.4, the above can be changed without too many
backward compatibility concerns.
Not applicable to 1.4 or below.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
matching word (e.g. "sip<TAB>"); this is implemented by this one-line change
- for (;; dst++, src += n) {
+ for (;src < argindex; dst++, src += n) {
However this code is not exactly trivial to understand, so
i am also adding some comments to help figuring out what it does.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Long explaination:
The behaviour of the underlying malloc(0) differs depending on the
operating system. Some return NULL (SysV behaviour); some still
allocate a small chunk of memory and return a valid pointer (e.g.
traditional BSD); some (e.g. FreeBSD 6.x) return a non-null pointer
that causes a memory fault if used, even just for reading.
Given the above variety, better never call malloc(0).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
are passed as an argument.
- Update the code in main/http.c to use the new interface
(the diff is large but mostly mechanical, due to the name change of
several variables);
- And since now it is trivial, implement "AMI over TLS", and document
the possible options in manager.conf
- And since the test client (openssl s_client -connect host:port )
does not generate \r\n as a line terminator, make get_input()
also accept just a \n as a line terminator (Mac users: do you
also need the \r-only version ?)
The option parsing in manager.conf is not very efficient, and needs
to be cleaned up and made similar to what we have in http.conf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
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r47986 | oej | 2006-11-24 07:00:19 -0700 (Fri, 24 Nov 2006) | 6 lines
Doxygen update
- Document cause codes
- Document a bit more on channel variables - global, predefined and local
- Fix some doxygen in channel.h. Adding one comment for two definitions does not
work. They won't be copied to each.
................
r47995 | murf | 2006-11-24 10:40:49 -0700 (Fri, 24 Nov 2006) | 1 line
This fix inspired by a patch supplied in bug 8189, which points out problems with the PLC code
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r47997 | murf | 2006-11-24 11:17:25 -0700 (Fri, 24 Nov 2006) | 1 line
removed the svnmerge-integrated property from trunk; it's confusing svnmerge in newly created branches
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r48001 | rizzo | 2006-11-25 02:02:42 -0700 (Sat, 25 Nov 2006) | 5 lines
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.
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r48003 | oej | 2006-11-25 02:45:57 -0700 (Sat, 25 Nov 2006) | 9 lines
- Adding comment on suspicious memory allocation. Seems like it's never freed, but I don't
have a clear understanding of the frame allocation/deallocation, so I just mark this
for investigation. (Reported by Ed Guy). We're trying to see if a free() hurts...
- Doxygen comments on p2p rtp bridge stuff. I am a bit worried about shortcutting
rtcp this way, but will need feedback from rtcp gurus. This should work for
video calls too, and possibly UDPTL.
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r48004 | oej | 2006-11-25 02:48:30 -0700 (Sat, 25 Nov 2006) | 2 lines
Changing ERROR to lesser level. Imported from 1.2/1.4
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r48008 | rizzo | 2006-11-25 10:37:04 -0700 (Sat, 25 Nov 2006) | 7 lines
generalize a bit the functions used to create an tcp socket
and then run a service on it.
The code in manager.c does essentially the same things,
so we will be able to reuse the code in here (probably
moving it to netsock.c or another appropriate library file).
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r48009 | mattf | 2006-11-25 13:30:04 -0700 (Sat, 25 Nov 2006) | 1 line
Updates to show linkset command
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r48010 | mattf | 2006-11-25 13:54:38 -0700 (Sat, 25 Nov 2006) | 2 lines
Add ss7 show linkset command
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r48011 | mattf | 2006-11-25 14:32:33 -0700 (Sat, 25 Nov 2006) | 1 line
Make sure we don't send a group reset on a group larger than 32 CICs
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r48012 | mattf | 2006-11-25 14:35:23 -0700 (Sat, 25 Nov 2006) | 1 line
bug fix
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r48013 | mattf | 2006-11-25 14:46:58 -0700 (Sat, 25 Nov 2006) | 1 line
Make compiler happier
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r48014 | mattf | 2006-11-25 14:50:42 -0700 (Sat, 25 Nov 2006) | 1 line
Little fix so we use the right message
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r48016 | murf | 2006-11-25 17:15:42 -0700 (Sat, 25 Nov 2006) | 9 lines
Merged revisions 48015 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1 line
A little bit of func_cdr documentation upgrade-- no bug# involved, although 8221 may have inspired it.
........
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r48018 | murf | 2006-11-25 17:31:13 -0700 (Sat, 25 Nov 2006) | 9 lines
Merged revisions 48017 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1 line
might as well also document the raw values of the flag vars
........
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r48019 | russell | 2006-11-25 23:55:33 -0700 (Sat, 25 Nov 2006) | 6 lines
- Add some comments on thread storage with a brief explanation of what it is
as well as what the motivation is for using it.
- Add a comment by the declaration of ast_inet_ntoa() noting that this function
is not reentrant, and the result of a previous call to the function is no
longer valid after calling it again.
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the infrastructure exposed in http.c earlier today.
As a bonus, now we can restart the session on a different
port just reloading the module.
On passing, fix a bug in the handling of 'enabled' in the configuration
file - previously, a missing "enabled=" line in manager.conf meant
"whatever the state was before" instead of a specific value.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
as described in
http://lists.digium.com/pipermail/asterisk-dev/2006-December/025213.html
In detail, this commit does the following:
b) change the function get_input() to use fread() instead of read()
to collect the data. One can still do the ast_wait_for_input() on
the original descriptor returned by accept().
c) change the function send_string() to work on the FILE *.
As a side effect, this change now really guarantees that
we don't spend more than "writetimeout" milliseconds on
each line sent.
d) modify the function action_command() so that it creates a
temporary file descriptor to be passed to ast_cli_command(),
and then read back the data from the temp file and write it
to the output with send_string(). The code is similar to
what is done in generic_http_callback() to support AMI-over-HTTP.
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and implement services over tcp and/or tcp-tls.
This commit is nothing more than moving structure definitions
(and documentation) from main/http.c to include/asterisk/http.h
(temporary location until we find a better place), and removing the
'static' qualifier from server_root() and server_start().
The name change (adding the ast_ prefix as a minimum, and then
possibly a more meaningful name) is postponed to future commits.
Does not apply to other versions of asterisk.
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Russell Bryant
Tilghman Lesher
Olle Johansson
Joshua Colp
Kevin P. Fleming
Jason Parker
Paul Cadach
Steve Murphy
Luigi Rizzo