DUNDi uses a concept called the Entity ID for unique server identifiers. I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this. The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf. DUNDi will now use this global EID unless one is specified
in dundi.conf.
The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the checks for the CLM and EVT services from the SAForum AIS. I'm going to work
on merging in changes from this branch in pieces.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines
Do not attempt to do emulation if an END digit is received and the length is
less than the defined minimum digit length, and the other end only wants END
digits (SIP INFO, for example).
(closes issue #12778)
Reported by: tsearle
Patches:
12778.rev1.txt uploaded by russell (license 2)
Tested by: tsearle
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(Note that this is being merged to trunk/1.6.0 because
it may affect non-callback agents with ackcall set)
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r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines
A unique situation of timeouts brought forth a failure situation for
autologoff in chan_agent. If using AgentCallbackLogin-style agents,
then if the timeout specified by the Dial() to reach the agent's phone
was shorter than the timeout specified in queues.conf, then autologoff
would only work if the caller hung up while the agent's phone was ringing.
This patch allows autologoff to work in this situation when the call in
queue transfers to the next available agent (as it would have if the timeout
in queues.conf were less than the timeout in the Dial()).
(closes issue #12754)
Reported by: Rodrigo
Patches:
12754.patch uploaded by putnopvut (license 60)
Tested by: Rodrigo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
where the entered phone number is checked.
You can now define a set of extensions/exten patterns that describe
valid phone numbers. PrivacyManager will check that context for a match
with the given phone number.
This way you get better control. For example people blindly hitting
10 digits just to get past privacymanager
Example line in extensions.conf:
exten => incoming,n,PrivacyManager(3,10,,route-outgoing)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.
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r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r120731 | russell | 2008-06-05 13:01:25 -0500 (Thu, 05 Jun 2008) | 2 lines
Add the UPGRADE.txt file from Asterisk 1.2, for handy reference.
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r120671 | russell | 2008-06-05 11:38:52 -0500 (Thu, 05 Jun 2008) | 5 lines
It turns out that searching on the forwarding station isn't very useful for
most people, so pull in the changes that allow searching for SMDI messages
based on other components of the SMDI message. Also, update the SMDI
documentation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
has been loaded, fix a return value in the loader, and ensure that the help
workhorse header does not print on load.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r120371 | russell | 2008-06-04 11:26:43 -0500 (Wed, 04 Jun 2008) | 4 lines
Make the "dialplan remove include" CLI command actually work. Also, tweak
some formatting, and make the success message a little bit more clear.
(closes AST-52)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) | 8 lines
Due to incorrect use of the AST_LIST_INSERT_HEAD() macro the loopback switch
cannot perform any translation on the extension number before searching for it
in the target context.
(closes issue #12473)
Reported by: chappell
Patches:
pbx_loopback.c.diff uploaded by chappell (license 8)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) | 6 lines
(closes issue #11594)
Reported by: yem
Tested by: yem
This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines
Fix another place where peer->callno could change at a very bad time, and also
fix a place where a peer was used after the reference was released.
(inspired by rev 120001)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a dynamic realtime queue member is added to the queue, and the
REMOVEMEMBER event is logged when a dynamic realtime member is
removed. Since no calling channel is associated with these events
the string "REALTIME" is placed where the channel's unique id is
normally placed.
(closes issue #12774)
Reported by: atis
Patches:
queue_log_rt_members.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
