- Avoid printing the name of an argument in the [Arguments] tag if there is no description
for that argument.
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add 1 to the priority when checking if the extension exists. Otherwise,
gosubs will fail.
This was discovered when investigating an asterisk-users mailing list post
made by Gary Hawkins.
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emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an
error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed
by Mark M. on IRC.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when a save occurs to a different filename, everything goes to the alternate
filename, instead of appending to the original. This is important for the
AMI command UpdateConfig.
(closes issue #13301)
Reported by: trevo
Patches:
20081113__bug13301.diff.txt uploaded by Corydon76 (license 14)
20081113__bug13301__1.6.0.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Terry Wilson created the original patch for this functionality, which I slightly modified and added
the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not
do anything for G711 over SIP fax detection. By default, this option is disabled.
Reviewboard: http://reviewboard.digium.com/r/69/
This functionality is for issue AST-140.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines
(closes issue #13835)
Reported by: matt_b
Tested by: jpeeler
This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.
Closes AST-142.
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r160943 | mmichelson | 2008-12-04 10:44:18 -0600 (Thu, 04 Dec 2008) | 15 lines
Fix a callerid parsing issue. If someone formatted callerid like the
following: "name <number>" (including the quotation marks), then the parts
would be parsed as
name: "name
number: number
This is because the closing quotation mark was not discovered since the number
and everything after was parsed out of the string earlier. Now, there is a check
to see if the closing quote occurs after the number, so that we can know if we
should strip off the opening quote on the name.
Closes AST-158
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We dont want to scare users with this, so we added a devmode compile flag
(closes issue #13952)
Reported by: wedhorn
Patches:
packetdebug3.diff uploaded by wedhorn (license 30)
Tested by: mvanbaak, wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r160764 | qwell | 2008-12-03 15:38:07 -0600 (Wed, 03 Dec 2008) | 7 lines
Only show this warning when we want to show it.
(closes issue #13982)
Reported by: coolmig
Patches:
chan_agent.c.patch uploaded by coolmig (license 621)
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r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | 11 lines
(closes issue #13597)
Reported by: john8675309
Patches:
patch.13597 uploaded by murf (license 17)
Tested by: murf, john8675309
This patch causes the setcid func to update the CDR
clid after setting the channel field.
I also notice that in trunk, the num/number of 1.4 is
left out; I decided to include the option to use
either in trunk, so as not to have 1.4 upgraders
not to have problems.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Move the code to start using the LIST macros.
Review: http://reviewboard.digium.com/r/72
(closes issue #13232)
Reported by: eliel
Patches:
iax2-provision.patch.txt uploaded by eliel (license 64)
(with minor changes pointed by Mark Michelson on review board)
Tested by: eliel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
for app_dial and app_queue to run a gosub when the call is answered.
* Check for the existence of the gosub target in gosub_exec. If it is nonexistent,
then this will cause errors when we attempt to actually run the gosub, including
a definite memory leak and potential crashes. Return an error in this situation
* Check the return value of pbx_exec in app_dial and app_queue before attempting
to actually run the gosub routine. If there was an error, we should not attempt
to run the gosub.
* Change a '|' to a ',' in app_queue.
* Add some extra curly braces where they had been missing previously.
(closes issue #13548)
Reported by: fiddur
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r160570 | tilghman | 2008-12-03 11:55:12 -0600 (Wed, 03 Dec 2008) | 5 lines
During bridge code, the channel bridge may return a retry code, if a transfer
was initiated but not yet completed. If the bridge is immediately retried,
then we may send a storm of TXREQ packets, even though the first set is sent
reliably (retransmitted). Fixes AST-137.
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r160558 | tilghman | 2008-12-03 11:34:34 -0600 (Wed, 03 Dec 2008) | 7 lines
If an entry is added to the directory during a scan when another entry expires,
then that new entry will not be processed promptly, but must wait for either a
future entry to start or a current entry's retry to occur. If no other entries
exist in the directory (other than the new entries) when a bunch expire, then
the new entries must wait until another new entry is added to be processed.
This was a rather weird race condition, really. Fixes AST-147.
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handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being
done on the incorrect channel.
With this set of changes, a gosub will correctly run on
the answering queue member's channel. There are still crash
issues which occur if there are dialplan syntax errors, so
I cannot yet close the referenced issue.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r160551 | tilghman | 2008-12-03 10:58:34 -0600 (Wed, 03 Dec 2008) | 4 lines
Don't start scanning the directory until all modules are loaded, because some
required modules (channels, apps, functions) may not yet be in memory yet.
Fixes AST-149.
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r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines
Jon Bonilla (Manwe) pointed out on the -dev list:
"I guess that having only ip-phones in mind is not a good approach. Since it is
possible to have a sip proxy connected to asterisk we could receive a 407
(unauthorized) or 483 (too many hops) as response and dialog ending would not be
a good behavior."
So modified.
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r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines
When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion
fails, and the resulting integer is garbage. Thus, we must initialize the
integer and check it afterwards for success.
(closes issue #14000)
Reported by: folke
Patches:
asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626)
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