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r76174 | qwell | 2007-07-20 15:32:55 -0500 (Fri, 20 Jul 2007) | 2 lines
It's possible for sub->owner to be NULL here if you cancel the call immediately after/during sending a digit.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76087 | file | 2007-07-20 14:20:09 -0300 (Fri, 20 Jul 2007) | 14 lines
Merged revisions 76080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6 lines
(closes issue #10247)
Reported by: fkasumovic
Patches:
chan_sip.patch uploaded by fkasumovic (license #101)
Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer.
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possible for it to be empty.
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because in this case the string is left-aligned and it is not
truncated anyways.
Omitting the field size prevents the generation of trailing whitespace,
which makes the string fit in smaller windows.
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localnet settings (requires the change in SVN 76034), and also
give an indication on whether/why/how the remapping of addresses
in SIP message is done or not.
I think this is especially useful for debugging the configuration,
as the address remapping depends on a combination of at least 3
parameters (localnet, externhost, externip) and successful DNS lookup.
An example of the output of this section is below:
Network Settings:
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SIP address remapping: Enabled using externhost
Externhost: foo.dyndns.net
Externip: 80.64.128.23:0
Externrefresh: 10
Internal IP: 12.34.56.78:5060
Localnet: 192.168.0.0/255.255.0.0
10.0.0.0/255.0.0.0
I leave to the community the judgement if the above info is a
useful addition for 1.4. It is not a bugfix, but it is neither a
new feature, only a useful diagnostic tool.
Note that I would like to move there also the bindaddress/port
information, in the usual addr:port format e.g.
Bindaddress: 0.0.0.0:5060
so that network information is all in one place.
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universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
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r75928 | russell | 2007-07-19 10:53:15 -0500 (Thu, 19 Jul 2007) | 14 lines
Merged revisions 75927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75927 | russell | 2007-07-19 10:49:42 -0500 (Thu, 19 Jul 2007) | 6 lines
When processing full frames, take sequence number wraparound into account when
deciding whether or not we need to request retransmissions by sending a VNAK.
This code could cause VNAKs to be sent erroneously in some cases, and to not
be sent in other cases when it should have been.
(closes issue #10237, reported and patched by mihai)
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in the "sip show settings" cli output. I have put these in a
separate section, probably even bindaddr and SIP port should go
there.
There are more things to add here e.g. localnet and so on.
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as well as the handling of the sip socket.
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and remove the code that would never be called.
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The l is useless.
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r75807 | qwell | 2007-07-18 17:59:18 -0500 (Wed, 18 Jul 2007) | 1 line
Need to make sure we set milliseconds and timestamp - pointed out by the recent ast_ time stuff from Tilghman
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list were not destroyed when the module is unloaded. However, after reading
the code related to the use of this list a lot today, I realized that it isn't
necessary. So, I have added a comment to explain why it isn't necessary.
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r75759 | russell | 2007-07-18 16:09:46 -0500 (Wed, 18 Jul 2007) | 13 lines
Merged revisions 75757 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75757 | russell | 2007-07-18 16:09:13 -0500 (Wed, 18 Jul 2007) | 5 lines
When traversing the queue of frames for possible retransmission after
receiving a VNAK, handle sequence number wraparound so that all frames that
should be retransmitted actually do get retransmitted.
(issue #10227, reported and patched by mihai)
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r75732 | qwell | 2007-07-18 15:17:27 -0500 (Wed, 18 Jul 2007) | 1 line
Umm, why are we transmitting dialtone on cfwdall?
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(closes issue #9245)
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r75711 | qwell | 2007-07-18 14:54:32 -0500 (Wed, 18 Jul 2007) | 4 lines
Fixes for 7935/7936 conference phones.
Issue 9245, patch by slimey.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #9887)
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r75707 | qwell | 2007-07-18 14:48:12 -0500 (Wed, 18 Jul 2007) | 4 lines
Fix issues with new 79x1 phones.
Issue 9887, patches by DEA
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microsecond, instead of only to the second
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r75623 | file | 2007-07-18 12:44:02 -0300 (Wed, 18 Jul 2007) | 2 lines
Few more places that needs to check for onhold state.
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r75621 | file | 2007-07-18 12:41:06 -0300 (Wed, 18 Jul 2007) | 5 lines
(closes issue #10165)
Reported by: elandivar
It is possible for hold status to exist without call limits set, so we need to ensure update_call_counter is executed regardless.
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r75619 | file | 2007-07-18 12:25:45 -0300 (Wed, 18 Jul 2007) | 2 lines
Don't bother reloading chan_h323 if it did not load successfully in the first place. This would otherwise cause a crash.
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where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc.
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didn't need to be checked in others.
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r75450 | russell | 2007-07-17 15:57:56 -0500 (Tue, 17 Jul 2007) | 11 lines
Merged revisions 75449 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75449 | russell | 2007-07-17 15:57:09 -0500 (Tue, 17 Jul 2007) | 3 lines
Properly check for the length in the skinny packet to prevent an invalid memcpy.
(ASA-2007-016)
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r75445 | russell | 2007-07-17 15:48:21 -0500 (Tue, 17 Jul 2007) | 13 lines
Merged revisions 75444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75444 | russell | 2007-07-17 15:45:27 -0500 (Tue, 17 Jul 2007) | 5 lines
Ensure that when encoding the contents of an ast_frame into an iax_frame, that
the size of the destination buffer is known in the iax_frame so that code
won't write past the end of the allocated buffer when sending outgoing frames.
(ASA-2007-014)
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r75441 | russell | 2007-07-17 15:42:12 -0500 (Tue, 17 Jul 2007) | 12 lines
Merged revisions 75440 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75440 | russell | 2007-07-17 15:41:41 -0500 (Tue, 17 Jul 2007) | 4 lines
After parsing information elements in IAX frames, set the data length to zero,
so that code later on does not think it has data to copy.
(ASA-2007-015)
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run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
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r75053 | russell | 2007-07-13 14:11:26 -0500 (Fri, 13 Jul 2007) | 20 lines
Merged revisions 75052 via svnmerge from
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r75052 | russell | 2007-07-13 14:10:00 -0500 (Fri, 13 Jul 2007) | 12 lines
(closes issue #9660)
Reported by: mmacvicar
Patches submitted by: bbryant, russell
Tested by: mmacvicar, marco, arcivanov, jmhunter, explidous
When using a TDM400P (and probably other analog cards) there was a chance that
you could hang up and pick the phone back up where it has been long enough to
be not considered a flash hook, but too soon such that the device reports that
it is busy and the person on the phone will only hear silence. This patch
makes chan_zap more tolerant of this and gives the device a couple of seconds
to succeed so the person on the phone happily gets their dialtone.
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r74955 | murf | 2007-07-12 14:42:08 -0600 (Thu, 12 Jul 2007) | 1 line
This patch resolves 10143; thanks to irroot for the patch; looked acceptable. Let the community decide if it messes things up
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r74866 | qwell | 2007-07-12 11:53:35 -0500 (Thu, 12 Jul 2007) | 1 line
It helps if I actually add this stuff for the 7921 too - otherwise it won't actually do much of anything.
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r74864 | qwell | 2007-07-12 11:48:49 -0500 (Thu, 12 Jul 2007) | 1 line
Add device ID for 7921 wireless skinny phone
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r74767 | russell | 2007-07-11 17:57:07 -0500 (Wed, 11 Jul 2007) | 13 lines
Merged revisions 74766 via svnmerge from
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r74766 | russell | 2007-07-11 17:53:26 -0500 (Wed, 11 Jul 2007) | 5 lines
The function make_trunk() can fail and return -1 instead of a valid new call
number. Fix the uses of this function to handle this instead of treating it
as the new call number. This would cause a deadlock and memory corruption.
(possible cause of issue #9614 and others, patch by me)
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r74722 | mmichelson | 2007-07-11 16:14:09 -0500 (Wed, 11 Jul 2007) | 13 lines
Merged revisions 74719 via svnmerge from
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r74719 | mmichelson | 2007-07-11 16:12:30 -0500 (Wed, 11 Jul 2007) | 5 lines
The cli command "agent logoff Agent/x soft" did not work...at all. Now it does.
(closes issue #10178, reported and patched by makoto, with slight modification for 1.4 and trunk by me)
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r74572 | file | 2007-07-11 14:03:08 -0300 (Wed, 11 Jul 2007) | 2 lines
Instead of figuring out kernel versions that have compiler.h and not... let's just use autoconf to check for it's presence. (issue #10174 reported by francesco_r)
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reported by snuffy)
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r74515 | file | 2007-07-11 11:09:13 -0300 (Wed, 11 Jul 2007) | 2 lines
Only check if we need to do a SIGMA based tone generation if we have a card. (issue #10179 reported by mikowhy)
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(closes issue #10169)
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r74379 | qwell | 2007-07-10 14:06:24 -0500 (Tue, 10 Jul 2007) | 12 lines
Merged revisions 74376 via svnmerge from
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r74376 | qwell | 2007-07-10 14:03:45 -0500 (Tue, 10 Jul 2007) | 4 lines
Fix an issue with wrapuptime not working when using AgentLogin.
Issue 10169, patch by makoto, with a minor mod by me to not re-break issue 9618
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for pointing this out (and pointing out what I probably did to prevent gcc from fixing it - don't ctrl-C builds)
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Closes issue #9186
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r74159 | qwell | 2007-07-09 15:19:28 -0500 (Mon, 09 Jul 2007) | 16 lines
Merged revisions 74158 via svnmerge from
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r74158 | qwell | 2007-07-09 15:18:15 -0500 (Mon, 09 Jul 2007) | 8 lines
Several chan_zap options were not working on reload because they were arbitrarily
disallowed when reloading some/most PRI options (such as signalling) was disallowed.
Options such as polarityonanswerdelay and answeronpolarityswitch can safely be changed on a reload.
This corrects that behavior.
Issue 9186, patch by tzafrir.
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r74082 | file | 2007-07-09 12:32:43 -0300 (Mon, 09 Jul 2007) | 2 lines
Only destroy the scheduler context if it was allocated. (issue #10124 reported by gzero)
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r74045 | file | 2007-07-09 11:49:05 -0300 (Mon, 09 Jul 2007) | 2 lines
Few minor thread synchronization tweaks. (issue #10124 reported by gzero)
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phones.
If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.
Thanks to Ramon and Frank for feedback on this feature.
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r73985 | tilghman | 2007-07-08 23:03:20 -0500 (Sun, 08 Jul 2007) | 2 lines
Doxygen formatting fixes; fixes errors while 'make progdocs'. (Closes issue #10104)
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r73849 | oej | 2007-07-08 11:47:31 +0200 (Sun, 08 Jul 2007) | 2 lines
While tracking down a bug, I need some more history. Dumphistory is very useful, indeed.
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r73769 | russell | 2007-07-06 18:02:58 -0500 (Fri, 06 Jul 2007) | 12 lines
Merged revisions 73768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | 4 lines
If a sip_pvt struct has already registered an extension state callback,
remove the old one before adding a new one. If this isn't done, Asterisk
will crash. (issue #10120)
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