Age | Commit message (Collapse) | Author | Files | Lines |
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r78569 | file | 2007-08-08 10:51:01 -0300 (Wed, 08 Aug 2007) | 4 lines
(closes issue #10335)
Reported by: adamgundy
Update sip.conf to include another scenario where directrtpsetup will fail.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78570 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Closes issue #10379, patch by mvanbaak.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78179 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #9779)
........
r77996 | qwell | 2007-08-02 16:53:39 -0500 (Thu, 02 Aug 2007) | 5 lines
Make sure we actually allow 6 chars to be sent.
Also make note of the "A" option of date format.
Issue 9779, modifications by DEA, wedhorn, and myself.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77997 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
method. "new" method being config file)
Add support for autocomplete of "say load" CLI command.
Patch by IgorG
(closes issue #10243)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76216 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@73298 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Issue 9762, patch by mvanbaak.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72741 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
and stop during message playback.
(closes issue 9474, reported and patched by jaroth with modifications by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72329 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
using IMAP storage.
This feature may be turned on by adding imapgreetings=yes to the general section of voicemail.conf
voicemail.conf.sample has details on the options added.
As a result, IMAP storage now has RETRIEVE and DISPOSE macros defined.
In addition to the IMAP greeting changes, I also have added an enum for the voicemail folders
and so now the code should be easier to understand and maintain when it comes to this area.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72232 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
This permission was discussed on the -dev mailing list some months back.
Issue 8613, patch by johann8384, with some minor changes by me.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70961 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
have cdr_custom capabilities. Many thanks to eserra for this contribution
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70122 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
with carriers that do Polarity/DTMF CID signalling.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@70001 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
SS7 number options (#10000)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69943 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Issue 9932, patch by eliel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68831 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68662 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
else seemed to think was very funny. Oh well ... :)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67895 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66856 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66818 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
disclaimer along with SIP messages in the header, X-Disclaimer. This is off by
default. Also, the text of the disclaimer can be customized in sip.conf.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66777 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
features.conf. This allows you to create a feature one time, and then map it
into groups for various different key mappings for the same feature, as well
as easy access control to groups of features.
(patch from bbryant)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66774 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66773 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66734 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
data is properly initialized.
(issue #9765, reported by MatsK, patch from eliel)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66724 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
class in musiconhold.conf, a caller on hold may press this digit to switch
to listening to that music class.
This involved adding a new callback for generators, which allow generators
to get notified of DTMF from the channel they are running on. Then, a callback
was implemented for the music on hold generators.
(patch from bbryant)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65505 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65169 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
manager user. (issue #8664, reported and original patch by ssokol, patch
updated by bkruse, and further updated by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64786 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
becomes unreachable
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64497 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64455 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64273 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
results.
Issue 9514.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64263 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Combined effort by DEA and mvanbaak.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64030 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r63329 | russell | 2007-05-07 17:28:50 -0500 (Mon, 07 May 2007) | 3 lines
Add a sample configuration file and example tables for use with res_config_pgsql.
(issue #9676, suretec)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63330 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r63047 | pari | 2007-05-04 11:45:29 -0500 (Fri, 04 May 2007) | 1 line
explanation for httptimeout in manager.conf
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63105 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(issue #7077, patch by adomjan)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62792 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
add some new options to control what happens when you hangup on an attended
transfer before the target extension answers the transferred channel. You
can now have it send the transferee back to the transferer.
(issue #8413, patch from sergee with very minor modifications by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62593 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r62497 | russell | 2007-05-01 11:26:48 -0500 (Tue, 01 May 2007) | 11 lines
Merged revisions 62496 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | 3 lines
Add indications.conf information for the Philippines.
(issue #9525, reported and patched by loloski)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62498 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62457 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r62371 | qwell | 2007-04-30 09:52:31 -0500 (Mon, 30 Apr 2007) | 2 lines
Remove unused (and potentially confusing) jitterbuffer options from sample config.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62372 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62292 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
control the
minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
(issue #9604, patch by Matthew Roth)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62242 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
blocks instead of one large monolithic app. Supports multiple templates
and is designed mostly for voicemail delivery over e-mail.
There's a todo with a list of ideas in the source code if you want
to contribute. Feedback is appreciated!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61671 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61324 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines
To be able to achieve the things that we would like to achieve with the
Asterisk GUI project, we need a fully functional HTTP interface with access
to the Asterisk manager interface. One of the things that was intended to be
a part of this system, but was never actually implemented, was the ability for
the GUI to be able to upload files to Asterisk. So, this commit adds this in
the most minimally invasive way that we could come up with.
A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in
the parser, and updated it to be thread-safe. The ability to check
permissions of active manager sessions was added by Dwayne Hubbard. Then,
hacking this all together and do doing the modifications necessary to the HTTP
interface was done by me.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60604 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r60323 | murf | 2007-04-05 16:35:11 -0600 (Thu, 05 Apr 2007) | 1 line
Added some clarification to the example configs for CDRs, on how to select a backend. Also, made cdr-csv the default if you 'make samples', and no other changes.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60324 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r59452 | murf | 2007-03-29 18:56:36 -0600 (Thu, 29 Mar 2007) | 1 line
A small clarification to keep bugs from being filed, and confusion from rising, if clearglobalvars is set, and globals are set in the AEL file. (9419)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59453 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r59040 | tilghman | 2007-03-19 10:42:26 -0500 (Mon, 19 Mar 2007) | 2 lines
Fix unescaped semicolon (reported via -dev list)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59041 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r58957 | russell | 2007-03-15 20:42:37 -0500 (Thu, 15 Mar 2007) | 1 line
fix a couple SLA documentation references
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58958 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) | 8 lines
By default, don't attempt to do any CallerID handling at all with SLA because
it is known to not work properly in some situations. However, add an option to
enable it for those that would like to use it anyway.
The short story behind this is that to properly handle CallerID with SLA, we
need the ability to change the CallerID on an existing call, and we are not
ready to handle that.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58895 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r58870 | russell | 2007-03-13 18:11:08 -0500 (Tue, 13 Mar 2007) | 1 line
fix the reference to the SLA documentation
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58871 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a
SQLite3 database. (issue #7149, alerios)
* Add new module, res_config_sqlite, which adds realtime database configuration
support for SQLite version 2. I decided that this was ok since we didn't have
any realtime support for version 3. If someone ports this to version 3, then
version 2 support can be removed or marked deprecated.
(issue #7790, rbarun_proformatique)
* Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom.
Also, note that there were other modules on the bug tracker that did not make
the cut because they provided some duplicated functionality. Those are:
* cdr_sqlite3 (issue #6754, moy)
* cdr_sqlite3 (issue #8694, bsd)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58866 f38db490-d61c-443f-a65b-d21fe96a405b
|