be stored in ast_flags. First victim is 'SIP_NO_HISTORY'
replaced by a 'do_history' field in the sip_pvt structure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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));
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75878 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r73679 | russell | 2007-07-06 10:57:25 -0500 (Fri, 06 Jul 2007) | 15 lines
Merged revisions 73678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | 7 lines
(closes issue #10125)
Reported by: makoto
Patches submitted by: makoto
This fixes a crash in chan_sip that happens when the bindaddr setting is not
valid on Asterisk startup, gets fixed, and then a reload gets issued.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69944 | russell | 2007-06-19 10:22:36 -0500 (Tue, 19 Jun 2007) | 10 lines
Fix a crash that could occur when handing device state changes.
When the state of a device changes, the device state thread tells the extension
state handling code that it changed. Then, the extension state code calls the
callback in chan_sip so that it can update subscriptions to that extension.
A pointer to a sip_pvt structure is passed to this function as the call which
needs a NOTIFY sent. However, there was no locking done to ensure that the pvt
struct didn't disappear during this process.
(issue #9946, reported by tdonahue, patch by me, patch updated to trunk to use
the sip_pvt lock wrappers by eliel)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69945 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69661 | file | 2007-06-18 11:46:32 -0400 (Mon, 18 Jun 2007) | 2 lines
Few minor transfer tweaks. We can't unlock something we never locked, and better handle a specific scenario with doing an attended transfer between two non-bridged calls.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69183 | russell | 2007-06-13 14:57:38 -0500 (Wed, 13 Jun 2007) | 9 lines
Move the logic for destroying a call when no response is received to a BYE
outside of the block that checks for FLAG_FATAL to be set. This flag is only
set when the packet is transmitted with the reliability set to XMIT_CRITICAL
when the original packet is transmitted. A BYE is always sent with it set
to XMIT_RELIABLE, meaning this code could never be encountered. This resulted
in seeing some SIP channels that would never go away with the last packet
sent being a BYE.
(part of issue #9235, patch from jcmoore)
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