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r58638 | russell | 2007-03-09 17:59:10 -0600 (Fri, 09 Mar 2007) | 8 lines
Merge some updates to the SLA documentation. I plan to keep working on this
to explain all of the expected behavior with call handling, configuration
details for specific phones, and other things. However, I got tired of doing
it in plain text, so I switched to using LaTeX. I have included the PDF version.
I haven't been able to get a nice looking plain text version out of it yet, but
I'm not terribly concerned since this is supposed to be more of the manual,
while the plain text sample configuration file is the reference.
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r58604 | file | 2007-03-09 16:08:19 -0500 (Fri, 09 Mar 2007) | 2 lines
Fix spelling of unavailable in voicemail documentation. (issue #9248 reported by tensai)
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r58584 | file | 2007-03-09 15:49:47 -0500 (Fri, 09 Mar 2007) | 10 lines
Merged revisions 58579 via svnmerge from
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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines
If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.
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r58512 | russell | 2007-03-08 16:15:15 -0600 (Thu, 08 Mar 2007) | 5 lines
Hang up the channel that put the call on hold in the event processing thread to
avoid a race condition. Also, if the station originated the call that it is
putting on hold, don't hang up the trunk if it was the only station on the call
and it is hanging up due to hold and not a normal hangup.
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r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) | 3 lines
Add a missing break statement so that handling the above event does not
incorrectly destroy the channel. (issue #9242, andrew)
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r58479 | tilghman | 2007-03-08 15:33:03 -0600 (Thu, 08 Mar 2007) | 2 lines
Fix segfault (Issue 9236)
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r58474 | russell | 2007-03-08 14:54:56 -0600 (Thu, 08 Mar 2007) | 3 lines
Refactor hold handling a bit so that it does not require keeping the call up
when a call is put on hold.
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r58436 | file | 2007-03-08 13:01:00 -0500 (Thu, 08 Mar 2007) | 2 lines
Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu)
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r58389 | file | 2007-03-08 11:07:10 -0500 (Thu, 08 Mar 2007) | 10 lines
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r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 lines
Only print out debug message if the definition that makes the variables shows up was actually defined. (issue #9233 reported by serginuez)
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r58354 | kpfleming | 2007-03-08 08:23:46 -0500 (Thu, 08 Mar 2007) | 2 lines
this change was not needed; fclose() handles closing the file descriptor already
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r58351 | kpfleming | 2007-03-08 08:17:17 -0500 (Thu, 08 Mar 2007) | 2 lines
fix two cases where HTTP session file descriptors would not be closed
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r58352 | kpfleming | 2007-03-08 08:17:42 -0500 (Thu, 08 Mar 2007) | 2 lines
fix a compiler warning, and overwriting 'res' value
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r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) | 6 lines
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
(issue #7256, tzafrir)
Also, update the configure script to make sure that we don't try to build
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.
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This can be done using a global variable or a dialplan function. Using the
SHELL() function will allow you to use an external script to determine what the
weight in the response should be. This can be very useful in load balancing
applications.
(inspired by discussions with blitzrage and jsmith in #asterisk-bugs)
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r58243 | russell | 2007-03-07 12:19:19 -0600 (Wed, 07 Mar 2007) | 17 lines
(This bug was reported to me by Kinsey Moore)
Merged revisions 58242 via svnmerge from
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines
Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call. This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication happens
since at that point, we have an exact match.
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r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines
Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)
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message.
(issue #9105, junky)
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r58165 | russell | 2007-03-06 18:25:19 -0600 (Tue, 06 Mar 2007) | 12 lines
Merged revisions 58164 via svnmerge from
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r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines
If the channels acquired using the manager Redirect action are not up, then
don't attempt to do anything with them. It could lead to weird behavior,
including crashes. (issue #8977)
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(inspired by issue #9215)
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addition to where it is already sent if either side hangs up.
(issue #9219, rgollent)
In passing, I put this code in a function so it would not be duplicated
a third time.
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r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue, 06 Mar 2007) | 9 lines
Merged revisions 58115 via svnmerge from
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line
Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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r58119 | russell | 2007-03-06 17:00:57 -0600 (Tue, 06 Mar 2007) | 3 lines
Clarify the documentation of the dialout and sendvoicemail options.
(issue #9000, caio1982 and serge-v)
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build here, too.
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r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue, 06 Mar 2007) | 10 lines
Merged revisions 58052 via svnmerge from
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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines
Change error message to proper message
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r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06 Mar 2007) | 3 lines
Return an error of transmit_response is called without a session.
(issue #9002)
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r57591 | russell | 2007-03-02 18:02:29 -0600 (Fri, 02 Mar 2007) | 1 line
add missing configuration template. Thanks to Lacy Moore on asterisk-users for pointing this out\!
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use directly using the ZT_GETVERSION ioctl. (issue #9094 reported by tootai)
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r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2 lines
Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh)
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r57872 | file | 2007-03-05 13:39:28 -0500 (Mon, 05 Mar 2007) | 2 lines
Don't create a listen channel and record the conference unless the option is turned on. (issue #9204 reported by francesco_r)
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r57870 | file | 2007-03-05 12:52:03 -0500 (Mon, 05 Mar 2007) | 10 lines
Merged revisions 57869 via svnmerge from
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r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines
Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares)
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r57826 | murf | 2007-03-05 08:20:17 -0700 (Mon, 05 Mar 2007) | 9 lines
Merged revisions 57825 via svnmerge from
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r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line
Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong)
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r57798 | file | 2007-03-04 23:19:53 -0500 (Sun, 04 Mar 2007) | 2 lines
Don't allow a NULL pointer to reach ast_frdup. (issue #9155 reported by cmaj)
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#9193 reported by junky)
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r57770 | file | 2007-03-04 22:35:03 -0500 (Sun, 04 Mar 2007) | 2 lines
Don't reference a potentially NULL pointer. (issue #9199 reported by klolik)
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r57768 | file | 2007-03-04 22:22:17 -0500 (Sun, 04 Mar 2007) | 2 lines
Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg)
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the conversion of IAX2 variables from the current custom method to ast_storage.
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r57649 | tilghman | 2007-03-03 00:45:00 -0600 (Sat, 03 Mar 2007) | 10 lines
Merged revisions 57648 via svnmerge from
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r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines
Memory leak of a list, if call recording was abandoned
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r57620 | dhubbard | 2007-03-02 18:59:24 -0600 (Fri, 02 Mar 2007) | 1 line
submitted patch for Georgian language, issue 9010, submitted by Alexander Shaduri
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Thanks to Lacy Moore on the asterisk-users list for pointing out that this
was missing!
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r57556 | russell | 2007-03-02 17:03:01 -0600 (Fri, 02 Mar 2007) | 3 lines
Update the check that is used to determine whether zaptel transcoder support
is present. The interface has changed.
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r57477 | file | 2007-03-02 12:06:52 -0500 (Fri, 02 Mar 2007) | 10 lines
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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines
If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.
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r57473 | murf | 2007-03-02 09:55:16 -0700 (Fri, 02 Mar 2007) | 9 lines
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r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 line
further refinement in wording of goto documentation, as per 9156, goto not proceeding to next instruction
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