Reported by: junky
Patches:
register_trying.diff.txt uploaded by jcmoore
Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82257 f38db490-d61c-443f-a65b-d21fe96a405b
+ 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: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
disclaimer along with SIP messages in the header, X-Disclaimer. This is off by
default. Also, the text of the disclaimer can be customized in sip.conf.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66777 f38db490-d61c-443f-a65b-d21fe96a405b
file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62457 f38db490-d61c-443f-a65b-d21fe96a405b
This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62292 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines
Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58780 f38db490-d61c-443f-a65b-d21fe96a405b
This replaces the older, broken, implementation where a setting in
[general] did not do anything and the [peer] part was broken.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53932 f38db490-d61c-443f-a65b-d21fe96a405b
Merged revisions 53109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines
Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53110 f38db490-d61c-443f-a65b-d21fe96a405b
If you set call limit and busy limit, chan_sip will indicate BUSY for a device
that has reached the busy limit and allow calls up to the call limit, allowing
for call transfers (that generate a new call).
If you only set call limit, chan_sip will not indicate BUSY until that limit
is filled.
This affects SIP subscriptions, call queues and manager applications.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53082 f38db490-d61c-443f-a65b-d21fe96a405b
- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48200 f38db490-d61c-443f-a65b-d21fe96a405b
to the peer side of a type=friend.
This is for trying to support BJ in his quest to solve some issues
with the queue system and type=friend objects.
BJ: Please test!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47201 f38db490-d61c-443f-a65b-d21fe96a405b
is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46489 f38db490-d61c-443f-a65b-d21fe96a405b
They are highly convenient for writing configuration files, especially
if you have many similar entries, or want to switch quickly between
different configurations without having to comment/uncomment large
sections of the files.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44579 f38db490-d61c-443f-a65b-d21fe96a405b
1. slightly rearrange/simplify the parsing of the argument in sip_register.
This brings in a patch that has been in Mantis (5834) for ages,
and is the larger part of the commit;
2. implement the "contact" option for peers, similar to the one in users.conf:
If you put a "contact" option with a non-empty argument (e.g. contact=123)
in a peer section, asterisk will register with the provider as if you had a
register= username:secret@host/contact
line in the general section.
The latter is a very small is a new feature so i am not putting it
in the 1.4 branch, although the "contact" option in user.conf is
already in the 1.4 branch and so it wouldn't be too strange to
merge it.
Note that the implementation of "contact" is much simpler than
the one in 5834, and limited to a few lines in build_peer().
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44566 f38db490-d61c-443f-a65b-d21fe96a405b
Technically, ooh323 doesn't support it yet, but there is a patch that should be committed very soon.
Issue #7989, patch by DEA, slightly modified.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43344 f38db490-d61c-443f-a65b-d21fe96a405b
dhubbard