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r128599 | mmichelson | 2008-07-07 09:35:27 -0500 (Mon, 07 Jul 2008) | 6 lines
Update a few instances of "extensions reload" to "dialplan reload"
in the documentation.
Patch provided by caio1982 (license 22)
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r128524 | oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines
- Fixing issues with "sip show settings"
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
binding to a different IP address
- Fixing documentation in sip.conf.sample
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r128237 | oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines
Make TCP disabled by default (it's considered experimental)
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r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2 lines
Reformatting the config sample
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r128122 | mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line
Add option to wait to be able to explicitly send ACM via the Proceeding() application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset
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r127154 | bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines
Add a configuration option so the global outboundproxy can use tcptls without it being defined by each sip user.
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r126845 | oej | 2008-07-01 14:54:57 +0200 (Tis, 01 Jul 2008) | 14 lines
Merged revisions 126844 via svnmerge from
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r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines
Clear up documentation on "domain=" setting in sip.conf
Reported by: davidw
(closes issue #12413)
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r126675 | jpeeler | 2008-06-30 17:34:08 -0500 (Mon, 30 Jun 2008) | 1 line
rename zapata.conf.sample to chan_dahdi.conf.sample
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r125891 | bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines
Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.
(issue #12799)
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r125223 | tilghman | 2008-06-25 20:25:16 -0500 (Wed, 25 Jun 2008) | 12 lines
Merged revisions 125218 via svnmerge from
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r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008) | 4 lines
Document ackcall=always.
(closes issue #12852)
Reported by: davidw
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r125191 | tilghman | 2008-06-25 20:11:43 -0500 (Wed, 25 Jun 2008) | 6 lines
Update sample configuration to match what are now the defaults for the prefix.
(closes issue #12838, related to issue #12198)
Reported by: pabelanger
Patches:
http.conf.diff2 uploaded by pabelanger (license 224)
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r123887 | tilghman | 2008-06-19 11:21:32 -0500 (Thu, 19 Jun 2008) | 12 lines
Merged revisions 123883 via svnmerge from
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r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines
Correct description of notifyringing option.
(Closes issue #12890)
Reported by gminet
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
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r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03 Jun 2008) | 5 lines
Move compatibility options into asterisk.conf, default them to on for upgrades,
and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
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r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines
Merged revisions 118646 via svnmerge from
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r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417)
Reported by: cstadlmann
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r118359 | tilghman | 2008-05-27 10:46:58 -0500 (Tue, 27 May 2008) | 11 lines
Merged revisions 118358 via svnmerge from
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r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008) | 3 lines
Add a note that pbx_config.so is needed for Local channels.
(Closes issue #12671)
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r117986 | tilghman | 2008-05-22 16:42:50 -0500 (Thu, 22 May 2008) | 2 lines
Add a compatibility option for upgrading realtime extensions
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r117792 | seanbright | 2008-05-22 11:49:17 -0400 (Thu, 22 May 2008) | 1 line
Minor text fix. roster -> resource.
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r116410 | qwell | 2008-05-14 15:43:26 -0500 (Wed, 14 May 2008) | 9 lines
Merged revisions 116409 via svnmerge from
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r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May 2008) | 1 line
Document exitcontext in app_voicemail sample config
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r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11 Apr 2008) | 3 lines
Make the sample config match the contributed LDAP schema
(Closes issue #12421)
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r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008) | 12 lines
Merged revisions 113874 via svnmerge from
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r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines
If the [csv] section does not exist in cdr.conf, then an unload/load sequence
is needed to correct the problem. Track whether the load succeeded with a
variable, so we can fix this with a simple reload event, instead.
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r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07 Apr 2008) | 2 lines
Additional note
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r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr 2008) | 1 line
Document 'originate' permission in manager sample config.
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r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) | 16 lines
Merged revisions 113118 via svnmerge from
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r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines
Allow playback with noanswer (and add earlyrtp option).
(closes issue #9077)
Reported by: pj
Patches:
earlyrtp.diff uploaded by wedhorn (license 30)
Tested by: pj, qwell, DEA, wedhorn
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r110691 | tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines
Update sample configurations to make virtual hosting more obvious.
(closes issue #11969)
Reported by: pprindeville
Patches:
acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)
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r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25 Mar 2008) | 6 lines
Update the sample configuration, to use Macro less (since it's now deprecated).
(closes issue #12293)
Reported by: pprindeville
Patches:
bugid-0012293.1.6.patch uploaded by pprindeville (license 347)
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r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines
Note that the TCP and TLS support is currently considered experimental and
is subject to change while we work out the remaining issues.
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r109775 | tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines
Change back to using ldap_initialize() and let the user specify a URL directly,
instead of trying to piece it together, badly.
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r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar 2008) | 10 lines
Add sample events for aastra phones.
aastra-check-cfg is the same as the other check-cfg entries,
and aastra-xml is to load a pre-configured xml script.
(closes issue #12229)
Reported by: gowen72
Patches:
aastra.patch uploaded by gowen72 (license 432)
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(closes issue #11587)
Reported by: sobomax
Patches:
dialstring_doc.diff uploaded by sobomax (license 359)
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(closes issue #12084)
Reported by: mmickan
Patches:
chan_vpb.cc.diff uploaded by mmickan (license 400)
module.h.diff uploaded by mmickan (license 400)
vpb.conf.sample uploaded by mmickan (license 400)
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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valid ciphers provided in both files. .. First commit since July, woot
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greater than two minutes). Remove
the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using
'1' as a queue-round-seconds value is no longer valid.
(closes issue #9736)
Reported by: caio1982
Patches:
queue_announce5.diff uploaded by caio1982 (license 22)
Tested by: caio1982, putnopvut
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r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb 2008) | 2 lines
improve 2BCT documentation a bit (thanks Jared)
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r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb 2008) | 2 lines
document usage of 'transfer' configuration option for ISDN PRI switch-side transfers
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r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008) | 3 lines
Clarify setting DYNAMIC_FEATURES so that it gets inherited by outbound channels.
(due to a discussion between me and a user via email)
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Closes issue #11875
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pre-release status.
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(closes issue #11875)
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r101219 | qwell | 2008-01-30 09:34:37 -0600 (Wed, 30 Jan 2008) | 5 lines
Change default config to use descending channel order of groups, rather than ascending.
Fixes a potential source of confusion in glare-type situations.
Issue 11875, reported by JimVanM.
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(closes issue #11797)
Reported by: macbrody
Patches:
app_rtppage-20080130.c uploaded by macbrody (license 352)
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drops packets that do not come from the remote party.
(closes issue #8952)
Reported by: amorsen
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(closes issue #5768)
Reported by: mguesdon
Patches:
res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121)
res_ldap.conf.sample uploaded by suretec (license 70)
asterisk-v3.1.4.ldif uploaded by suretec (license 70)
asterisk-v3.1.4.schema uploaded by suretec (license 70)
Tested by: oej, mguesdon, suretec, cthorner
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from cli.conf
after a discussion on the -dev list.
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in queues.conf.
Suggestion comes from Shaun2222 on IRC.
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