(I don't have this building on this machine, so caio1982 on IRC is going to
test it for me. :) )
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96743 f38db490-d61c-443f-a65b-d21fe96a405b
revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96717 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008) | 7 lines
Fix the problem of notification of a device state change to a device with a '-'
in the name. Could probably do with a better fix in trunk, but this bug has
been open way too long without a better solution.
Reported by: stevedavies
Patch by: tilghman
(Closes issue #9668)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96576 f38db490-d61c-443f-a65b-d21fe96a405b
This authentication mechanism is implemented under the iksemel API, which
makes use of GnuTLS, whereas we use OpenSSL. Note : there's ongoing
dicsussion at the SASL IETF WG in order to deprecate SASL DIGEST-MD5, see
http://ietfreport.isoc.org/ids-wg-sasl.html.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96499 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | 7 lines
Make use of the temporary channel pointer while the pvt is unlocked.
(closes issue #11675)
Reported by: flefoll
Patches:
chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96450 f38db490-d61c-443f-a65b-d21fe96a405b
remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96245 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan 2008) | 4 lines
We need to reset the membername to NULL on each iteration of this loop, otherwise the result is that
multiple members can have the same name, since the variable was not reset on each iteration of the loop.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96103 f38db490-d61c-443f-a65b-d21fe96a405b
of a hack. It just asks the core to generate the same tone that it would when
you hear ringback when making an outbound call. But hey, it works, and you get
the localized ring tone for the appropriate language set on the channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96079 f38db490-d61c-443f-a65b-d21fe96a405b
........
r96024 | russell | 2008-01-02 16:14:28 -0600 (Wed, 02 Jan 2008) | 2 lines
Convert locks of the contexts list in pbx_config to the appropriate rdlock or wrlock
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96025 f38db490-d61c-443f-a65b-d21fe96a405b
........
r96022 | russell | 2008-01-02 16:04:47 -0600 (Wed, 02 Jan 2008) | 2 lines
pbx_dundi only needs a rdlock on the contexts list.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96023 f38db490-d61c-443f-a65b-d21fe96a405b
........
r96020 | russell | 2008-01-02 16:00:21 -0600 (Wed, 02 Jan 2008) | 2 lines
app_macro only needs a rdlock on the contexts list.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96021 f38db490-d61c-443f-a65b-d21fe96a405b
(this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96019 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4 lines
Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-001)
(closes issue #11637)
Reported by: greyvoip
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95947 f38db490-d61c-443f-a65b-d21fe96a405b
to use AST_STANDARD_APP_ARGS instead of AST_NONSTANDARD_APP_ARGS for determining
member data.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95945 f38db490-d61c-443f-a65b-d21fe96a405b
finish implementation of check for Zaptel HWGAIN support
add check for Zaptel ECHOCANCEL_PARAMS support
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95937 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan 2008) | 9 lines
A change to improve the accuracy of queue logging in the case where a member does not
answer during the specified timeout period. Prior to this change, there was a small chance
that the member name recorded in this case would be blank. Also prior to this change, if using
the ringall strategy, if no one answered the call during the specified timeout, the member name
listed in the queue log would randomly be one of the members that was rung.
(closes issue #11498, reported and tested by hloubser, patched by me)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95891 f38db490-d61c-443f-a65b-d21fe96a405b
not work for everyone, but it did for some. This set of changes makes trunk
start again for those having problems. Instead of building libresample as a
static library, it just links the object files in directly with the asterisk
binary.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95864 f38db490-d61c-443f-a65b-d21fe96a405b
rizzo