+ 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));
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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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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