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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r76227 | russell | 2007-07-20 21:02:54 -0500 (Fri, 20 Jul 2007) | 12 lines
Merged revisions 76226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) | 4 lines
Backport a fix for a memory leak that was fixed in trunk in reivision 76221
by rizzo. The memory used for the localaddr list was not freed during a
configuration reload.
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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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r76178 | qwell | 2007-07-20 16:03:57 -0500 (Fri, 20 Jul 2007) | 7 lines
Allow getting a call from an existing "sub" channel.
Cancel ringing if endpoint hangs up before answering.
Fixes were backported from trunk (there was apparently a bit of confusion during merge of a previous patch).
(closes issue #10241)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76139 | mmichelson | 2007-07-20 13:42:27 -0500 (Fri, 20 Jul 2007) | 6 lines
When using users.conf for the entries in the directory, if multiple users had the same last name, only the first user listed would be available
in the directory.
(closes issue #10200, reported by mrskippy, patched by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76132 | russell | 2007-07-20 13:22:24 -0500 (Fri, 20 Jul 2007) | 6 lines
Use the define that specifies the default length of an artificially created
DTMF digit in the ast_senddigit() function. The define is set to 100ms by
default, which is the same thing that this function was using. But, using
the define lets changes take effect in this case, as well as the others where
it was already used.
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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:
---------------------------
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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(e.g. chan_sip.c in a subsequent commit).
Obviously exposing the internals of a data structure is far from ideal
(especially in a case like this where the implementation is very
inefficient and will need to be changed at some point).
On the other hand, it was also unclear what additional APIs should
we provide instead, and because exposing the stucture has no impact
on source and binary compatibility, this seemed to me the best option at
this time.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r75969 | mmichelson | 2007-07-19 11:26:10 -0500 (Thu, 19 Jul 2007) | 10 lines
Changes in handling return values of several functions in app_queue. This all started as a fix for issue #10008
but now includes all of the following changes:
1. Simplifying the code to handle positive return values from ast API calls.
2. Removing the background_file function.
3. The fix for issue #10008
(closes issue #10008, reported and patched by dimas)
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This merges the trunk only part of the patches from this issue. In 1.4, res_agi
will issue a warning if you try to use DeadAGI on a channel that is not hung up.
Now, in trunk, it just plain won't let you do it.
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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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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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