Age | Commit message (Collapse) | Author | Files | Lines |
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you can now use it with zaptel channels. (done while in Montreal at the Asterisk bootcamp!)
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(closes issue #10711, rbraun_proformatique)
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Reported by: junky
Patches:
register_trying.diff.txt uploaded by jcmoore
Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r82091 | mmichelson | 2007-09-10 10:02:12 -0500 (Mon, 10 Sep 2007) | 5 lines
Removing non-existent options from misdn configuration sample.
(closes issue #10678, reported and patched by IgorG)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81886 | mmichelson | 2007-09-07 10:25:19 -0500 (Fri, 07 Sep 2007) | 3 lines
Moving the explanation for joinempty to a more appropriate place
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10644)
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r81453 | qwell | 2007-09-04 14:56:06 -0500 (Tue, 04 Sep 2007) | 4 lines
Change default followme config file to point to the correct files.
Issue 10644, patch by pabelanger
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Reported by: pabelanger
Patches:
extensions.ael.sample.patch uploaded by pabelanger (license 224)
Update extensions.ael.sample with voicemail and | changes.
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regarding setinterfacevar.
MEMBERREALTIME indicates whether a member is realtime.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81379 | russell | 2007-08-30 10:33:48 -0500 (Thu, 30 Aug 2007) | 3 lines
Fix a typo, update a reload command, and remove an unused configuration file.
(closes issue #10606, casper)
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issue #10398)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81226 | russell | 2007-08-28 10:41:15 -0500 (Tue, 28 Aug 2007) | 2 lines
Add Russian tones. (closes issue #7953, hanabana)
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Reported by: IgorG
Patches:
sip_conf-80933-1.patch uploaded by IgorG (license 20)
Fix up sip.conf sample configuration.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80130 | qwell | 2007-08-21 10:03:45 -0500 (Tue, 21 Aug 2007) | 7 lines
(closes issue #10510)
Reported by: casper
Patches:
cdr.conf.diff uploaded by casper (license 55)
Fix a few errors in sample cdr config file.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80047 | qwell | 2007-08-20 11:08:49 -0500 (Mon, 20 Aug 2007) | 7 lines
(closes issue #10499)
Reported by: casper
Patches:
extensions.conf.sample.diff uploaded by casper (license 55)
Update CLI examples in extensions.conf.sample to reflect command changes.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78951 | file | 2007-08-10 10:49:19 -0300 (Fri, 10 Aug 2007) | 4 lines
(closes issue #10422)
Reported by: bhowell
Add note to sample configuration about module load order and how it can cause perfectly good queue members to be marked as invalid.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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Closes issue #10379, patch by mvanbaak.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #9779)
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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.
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+ 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));
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method. "new" method being config file)
Add support for autocomplete of "say load" CLI command.
Patch by IgorG
(closes issue #10243)
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Issue 9762, patch by mvanbaak.
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and stop during message playback.
(closes issue 9474, reported and patched by jaroth with modifications by me)
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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.
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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
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have cdr_custom capabilities. Many thanks to eserra for this contribution
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with carriers that do Polarity/DTMF CID signalling.
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SS7 number options (#10000)
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Issue 9932, patch by eliel
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else seemed to think was very funny. Oh well ... :)
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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.
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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)
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data is properly initialized.
(issue #9765, reported by MatsK, patch from eliel)
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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)
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manager user. (issue #8664, reported and original patch by ssokol, patch
updated by bkruse, and further updated by me)
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becomes unreachable
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results.
Issue 9514.
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Combined effort by DEA and mvanbaak.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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