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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76803 | qwell | 2007-07-24 11:32:20 -0500 (Tue, 24 Jul 2007) | 3 lines
Don't create the Asterisk channel until we are starting the PBX on it.
(ASA-2007-018)
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This allows you to just Dial(Skinny/line), as long as line isn't ambiguous.
Note that this does not remove or deprecate the "old" syntax, as it's still
quite useful - even moreso if shared lines get implemented.
Initial patch by me, with some changes and suggestions from wedhorn.
(closes issue #10263)
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+ externexpire was not initialized properly;
+ stunaddr was not handled properly on a sip reload
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applications
using old methods of parsing arguments to using the standard macros. However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76656 | qwell | 2007-07-23 13:59:28 -0500 (Mon, 23 Jul 2007) | 3 lines
Fix some incorrect softkey labels in messages.
Don't try to play dialtone in some unimplemented features.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76654 | file | 2007-07-23 15:29:48 -0300 (Mon, 23 Jul 2007) | 12 lines
Merged revisions 76653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4 lines
(closes issue #5866)
Reported by: tyler
Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10276)
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r76620 | qwell | 2007-07-23 12:57:53 -0500 (Mon, 23 Jul 2007) | 4 lines
Don't try to queue up hold/unhold frames on a non-existent channel.
Issue 10276.
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does not use DTMF BEGIN frames.
1.4 seems correct (it does not have the two fields).
However, as this bug shows, the current way of creating the sip_tech
replica is too error-prone, one can easily forget to update one of
the two entries. Perhaps it would be better to create sip_tech_info
expliclty at module load, by doing
sip_tech_info = sip_tech;
sip_tech_info.send_digit_begin = NULL
(in this case, this is something applicable to 1.4 as well).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76561 | file | 2007-07-23 11:34:21 -0300 (Mon, 23 Jul 2007) | 14 lines
Merged revisions 76560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 lines
(closes issue #10236)
Reported by: homesick
Patches:
rpid_1.4_75840.patch uploaded by homesick (license 91)
Accept Remote Party ID on guest calls.
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allow building under dev mode
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Reported by: snuffy
Patches:
doxygen-updates.diff uploaded by snuffy (license 35)
Another big batch of doxygen documentation updates
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between integers and strings using a single translation table,
and use them in a few places instead of ad-hoc routines
that duplicate the table.
On passing, note that REFER_CONFIRMED is never used, and add a
few comments.
Nothing to backport here.
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now also use other macros if present.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76519 | file | 2007-07-23 10:23:09 -0300 (Mon, 23 Jul 2007) | 6 lines
(closes issue #10268)
Reported by: mvanbaak
Patches:
chan_skinny_openbsd.diff uploaded by mvanbaak (license 7)
Add another OS that has to use the Macros for byte ordering.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76485 | russell | 2007-07-23 07:25:01 -0500 (Mon, 23 Jul 2007) | 6 lines
Use a signed integer for storing the number of bytes in the packet read from
the network. Using an unsigned value here made it impossible to handle an
error returned from recvfrom(). Furthermore, in the case that recvfrom()
did return an error, this would cause a crash due to a heap overflow.
(closes issue #10265, reported by and fix suggested by timrobbins)
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handling of the sip_register structures.
This commit only changes comments and whitespace.
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using the new-style cli format.
No functional changes, nothing to backport.
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foo = sip_destroy(foo);
and reduce the chance of bugs due to dangling pointers.
Also remove a duplicate prototype for the function.
nothing to backport.
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of an 'authpeer' local variable.
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responses, so that there is a common exit point.
Mark two places where probably we could return -1 instead of 0 to report
an error to the caller.
(change triggered by investigations on how the 'SIP_PKT_IGNORE' field was used).
nothing to backport from this commit
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SIP_PKT_* stuff from the previous commit.
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individual variables. Apart from SIP_PKT_IGNORE which was used
a zillion times, the other two are used seldom.
On passing:
- move the arrays to the end of struct sip_request, so a (small)
buffer overflow is less likely to overwrite the other fields;
- note that the 'ignore' argument to handle_invite_replaces() is not
used and should be removed (will be done in a separate commit).
Nothing to backport in this change.
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Also correct some items that were misclassified.
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variables and not flags.
NOTE:
The old behaviour (preserved in this commit) is that if sipdebug
is set in the config file, it can only be disabled by reloading the
config. I am not sure if this is accidental or voluntary, but it
is really unconvenient and I think it should be handled in the same
way as other options i.e. consider requests from the config file
or the cli (or the command line) to be fully equivalent and act on
the same status variable.
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before using it.
I am unclear on the details right now so i hope someone can comment
more. The obvious (and lazy) approach would be to bzero() all of it
(except for the string pool), but isn't that too much work ?
Feedback wanted here...
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(spotted by a diff with my own branch)
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be stored in ast_flags. First victim is 'SIP_NO_HISTORY'
replaced by a 'do_history' field in the sip_pvt structure.
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the sdp messages. Overall the code is slightly more readable
(because the string is fully described by a single pointer),
and more efficient (because the length is stored explicitly
so you don't need to do strlen()).
(I have been using this code for almost a year now.)
I wish we had infix string operators to do this sort of things!
Nothing to backport from this change.
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define and use a macro to determine whether we are pointing to
one of them, so when one goes away (or a new one appears) we don't
have to touch all the code.
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+ extensive documentation changes both in sip.conf.sample and in the source;
+ allow "externip" and "externhost" to include a port number as well;
+ allow "bindaddr" to have a port number (making bindport unnecessary,
even though it is still present for backward compatibility);
+ introduce the new "stunaddr" parameter to specify an STUN server to
be used from the main SIP socket;
+ extend the "sip show settings" output to show all the above.
Internally:
+ change related data structures from struct in_addr to struct sockaddr_in
to store the port numbers as well;
+ reorganize ast_sip_ouraddrfor() (should also be renamed to sip_ouraddrfor()
because it is not a generic API, though it might become so if called with
a socket as an additional argument, in which case it can be moved elsewhere).
As mentioned in the documentation, media sessions still do not use STUN so the
port numbers may still be incorrect when Asterisk is behind a NAT
On passing, some of the debugging messages printing media addresses are
probably using the wrong values, but this will be checked/fixed in a
subsequent commit if needed.
Part of the following chunk in the function that handles a "sip reload" is
probably needed on previous versions as well, to avoid leaking the memory
used for the "localaddr" list:
@@ -17244,13 +17274,17 @@
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
+ memset(&stunaddr, 0, sizeof(stunaddr));
+ memset(&internip, 0, sizeof(internip));
+ /* Free memory for local network address mask */
+ ---> ast_free_ha(localaddr); <-----
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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