f5e5a443cf
Changed occurrences of "busy-level" to "busylevel" in sip.conf.sample
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in light of commit 89441. Thanks to pj for pointing out the need for this
(closes issue #11307 , reported by pj)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
eab6b00904
Add support for application/dtmf SIP INFO dtmf handling. Yep, another
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way of handling DTMF in SIP. Totally undocumented, but implemented
in enough devices so we have to support it.
Code by sergee, small changes by oej.
Closes issue #11049
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
0f53904918
merged jcmoore's patch for configurable SDP origin-field username and session field, closes issue# 10795
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
a9c2f441d3
Merged revisions 82751 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(Closes issue #10753 )
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r82751 | qwell | 2007-09-18 10:28:21 -0500 (Tue, 18 Sep 2007) | 4 lines
Correct the allowexternaldomains option in SIP sample config.
Issue 10753
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
9bd4b3e353
Lil' bit more documentation to keep folks happy.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
9642d93117
(closes issue #9433 )
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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: https://origsvn.digium.com/svn/asterisk/trunk@82257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
7c760f67c3
(closes issue #10569 )
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Reported by: IgorG
Patches:
sip_conf-80933-1.patch uploaded by IgorG (license 20)
Fix up sip.conf sample configuration.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@80962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
afceb3e4aa
Merged revisions 78569 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78569 | file | 2007-08-08 10:51:01 -0300 (Wed, 08 Aug 2007) | 4 lines
(closes issue #10335 )
Reported by: adamgundy
Update sip.conf to include another scenario where directrtpsetup will fail.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
2286afa3af
Enhance NAT support as discussed on the -dev list, i.e.:
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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
19 years ago
cb55dbe8eb
Update documentation for proper CLI commands. (issue #9936 reported by eserra)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
6aec360466
Remove our little joke that was making fun of email disclaimers which nobody
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else seemed to think was very funny. Oh well ... :)
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19 years ago
0b6c6b2e89
Add some more information about the SIP Disclaimer header.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
3ce231fe95
fix a typo.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
3d2b58751f
To satisfy some legal concerns, add an option for chan_sip to include a
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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: https://origsvn.digium.com/svn/asterisk/trunk@66777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
90bad9d2f5
Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
b419fc1134
Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
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file doc/qos.tex has been updated to document the new functionality.
(issue #9540 , patch submitted by IgorG)
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19 years ago
b6b1bf3213
Merge changes from team/russell/events
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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: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
ea226e9d77
Merged revisions 58779 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
88928f67ed
Make documentation match the source code.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
32495f91f0
Add support for outbound proxy for peers and [general]
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This replaces the older, broken, implementation where a setting in
[general] did not do anything and the [peer] part was broken.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
44c6630e4d
rename busy-limit to busy-level, since it is not a limit
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actually parse the busy-limit option from sip.conf, instead of ignoring it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
cfe66e6b26
Patch based on this patch with small changes for trunk...
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Merged revisions 53109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
0b84b386b9
Implementing "busy-limit".
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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: https://origsvn.digium.com/svn/asterisk/trunk@53082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
064e6cff1a
Merged revisions 53062 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53062 | oej | 2007-02-01 17:35:12 +0100 (Thu, 01 Feb 2007) | 2 lines
Add explanation of port= in combination with defaultip= (thanks jsmith)
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19 years ago
0375227e5c
Added some docs
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
29ed493b40
Be politically correct
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
da7a35a1cc
Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
d1b621c6a5
Adding docs on t.38
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19 years ago
c23bc46089
- Disable RTP timeouts during T.38 transmission
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- 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.
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19 years ago
4ce5b7c080
- Remove T.38 early media, since T.38 requires two way communication (imported from 1.4)
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- Small fixes to limitonpeer
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
c946e3b3fb
Merged revisions 48143 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines
Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage)
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19 years ago
7e46275b51
Clarify some settings for status reports in subscriptions, queues and manager
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19 years ago
e5145bebe4
Explain RTP timeouts
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
4e47ce525b
Update docs for videosupport
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19 years ago
a6f5adefa1
Make it possible to enable/disable onhold tracking, in order to make life easier
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for realtime users.
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19 years ago
a427a2a89a
- CANCEL never uses authentication
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- Add docs on canreinvite
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19 years ago
d900b47ccf
Adding new config option "limitpeersonly" to only apply call limits
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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!
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19 years ago
b136baaff4
Fix rport handling.
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...where did the 1.2 properties come from, really? they're back.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f98f457727
Change name of "contact" setting to "callback" which better reflects what it
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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.
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19 years ago
e85d8e98d1
document the match_auth_username option
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19 years ago
a8a26ad389
Update of docs
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19 years ago
c62784c10d
In the course of a data this has been turned into an option to ignore replies, then ignore responses and finally I'm just getting rid of the option altogether and making it the default no matter what. C'est la vie!
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19 years ago
da330feb60
Merged revisions 45280 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45280 | file | 2006-10-16 16:06:18 -0400 (Mon, 16 Oct 2006) | 10 lines
Merged revisions 45265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2 lines
Use responses rather then replies even though they mean the same thing.
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19 years ago
b58cc9e1bd
Merged revisions 45262 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45262 | file | 2006-10-16 15:37:34 -0400 (Mon, 16 Oct 2006) | 10 lines
Merged revisions 45260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2 lines
Add 'ignoreoodreplies' option which will not create a pvt structure on a SIP response but instead basically drop it.
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19 years ago
77c69dc4ef
Recommend using "sip reload" since it's much easier to learn and
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remember.
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19 years ago
b19b4b9764
document a bit the use of templates.
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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.
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19 years ago
f94849ca2a
document the "contact" option a bit better.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
ccca5843fd
Two things:
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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().
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19 years ago
2a7ac3f735
update example commands to match current syntax
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(does not apply to 1.4)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
8bd82ebc0d
Add documentation on rtp packetization.
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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: https://origsvn.digium.com/svn/asterisk/trunk@43344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
091e1aed8d
Merged revisions 42716 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42716 | tilghman | 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) | 2 lines
Spelling/grammar fixes (Issue 7929)
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19 years ago
c6977b9983
Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
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19 years ago
6d0742fc16
merge Russell's 'hold_handling' branch, finally implementing music-on-hold handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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20 years ago
4376af0080
actually make the non-standard G726-32 behavior available for SIP clients
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0e0059c0f3
Remove configuration option "restrictcid" that is nowhere to
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be seen in the code. Did it exist, was it planned to exist
or was it documentationware only? Ask Dr Asterisk.
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20 years ago
b971f65978
- Make use of system name in realtime SIP peers optional
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- Fix small issue with SIP history
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20 years ago
f3594bd1a0
Removing configuration options that does not do anything yet. No need to
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add "promises" to the sip.conf.sample...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
dec3d7d4c0
Merged revisions 36253-36254 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r36253 | kpfleming | 2006-06-29 02:19:27 -0500 (Thu, 29 Jun 2006) | 2 lines
add documentation for peer-specific 'outboundproxy' setting
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r36254 | kpfleming | 2006-06-29 02:19:54 -0500 (Thu, 29 Jun 2006) | 2 lines
clarify documentation for 'persistentmembers' setting
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20 years ago
4177596e8d
reformatting sip.conf.sample a bit, adding dumphistory that was not documented
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
cc43f0bdc7
Speling error. Avoid swenglish :-) (thanks, jtodd!)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
e2b0c5b558
Add example of permit/deny to sip.conf.sample
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5456f425c6
Allow AST_FRAME_MODEM frames to be dumped, and document T.38 passthrough support
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4c76028de9
- add the ability to configure forced jitterbuffers on h323, jingle,
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and mgcp channels
- remove the jitterbuffer configuration from the pvt structures in
the sip, zap, and skinny channel drivers, as copying the same global
configuration into each pvt structure has no benefit.
- update and fix some typos in jitterbuffer related documentation
(issue #7257 , north, with additional updates and modifications)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6bce269454
Merged revisions 31321 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r31321 | kpfleming | 2006-06-01 07:41:47 -0500 (Thu, 01 Jun 2006) | 2 lines
remove a sample entry that never should have been added (code to support it was not merged)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
bb7dd96cfe
Add support for using a jitterbuffer for RTP on bridged calls. This includes
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854 , Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3e99be68d1
add a new option for 'obscuring' SIP user/peer names from fishers
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use an enum for authentication results and clean up code
fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
42cf0b0a8f
add another media path reinvite 'flavor', where we will only redirect our media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them)
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also, documented the 'canreinvite=update' option in the sample config file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6d603ec09c
Allow contexts in regexten so that extensions can be added to multiple contexts when peer registers (issue #6869 reported by and created by Marquis)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5237a0e06d
- Use systemname for realm in sip, if we have no configuration for realm
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- Optionally send systemname in manager (cool when you have a manager proxy)
- Use systemname in CLI prompt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
ca6cf552f9
Add documentation on "allowtransfer"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
7bbb6bd3aa
- fix typo in rtp.c, devicestate.h
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- add information about subscriptions and realtime dial plans in sip.conf.sample
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
c38fbd246e
note that group assignments must be from 0 to 63 (issue #7048 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5873462c2e
- Add doxygen documentation for sipsock_read locking
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- Improve documentation of pedantic
(related to issue #7016 )
From the air above Russia...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
023e27f695
Formatting fixes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
95de51526a
Added information on call-limit and realtime
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8410e0d681
support subscription-based MWI, and use proper Call-ID on NOTIFY messages (issue #6390 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
278b8e8fc7
improve IP TOS support for SIP and IAX2 (issue #6355 , code from jcollie plus modifications)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
83d9331261
Issue #5427
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- Enable videosupport per device
- Implement maxcallbitrate setting for video calls
Patch by John Martin, thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
18de2b7787
Issue #6705 (oej)
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- Implement option for allow/disallow subscriptions
- Implement option for allow/disallow overlap dialling
- Set default to disable overlap dialling in sip.conf.sample for new installations
- Remove overlap dialling from subscription logic
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
d7b5a18f4c
Fix reference to README files
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
1a206c1bf8
Clarify documentation for "progressinband" - imported from 1.2
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6b8701cffa
Whitespace changes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b40bd71a9a
restore 'rfc2833' naming for DTMF mode in chan_sip
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4d07b89fdd
- Change "rfc2833" to "rtp" in sip.conf. Keeping backwards compatibility.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3f6cc5c544
- Clarify default setting of canreinvite (thanks royk)
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- Add some extra headers and reference to other doc/ files for realtime
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
125fd8446c
Issue 5892: Set a minimum T1 timer for calls. Reporter: twisted
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b64404e039
From now on, apply maxexpiry and minexpiry to all subscriptions. Thanks to fourcheeze in the IRC channel
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for pointing this out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0ba27e0a6b
Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug #6183 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4dc76fbcc1
Fix comments in sip.conf ( #6134 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
125db028c3
- Add DOC file about caller ID presentation values
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- Add callingpres to sip.conf
- Add reference to README.callingpres from zapata.conf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8336 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5462ec082c
- Remove "incominglimit" as a configuration option in sip.conf
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- Add documentation on call-limit, explaining that there's two counters
for a type="friend".
- Document the removval of "incominglimit" in UPGRADE.txt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3b4f660a85
Bug 5345; Add configuration option for minimum registration time. (folsson)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b60daeb58f
Merged revisions 7599 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7599 | russell | 2005-12-22 15:36:47 -0500 (Thu, 22 Dec 2005) | 3 lines
revert changes to videosupport to allow per-peer setting, since it isn't quite
complete and there is not an obvious fix at this point
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b6b32ebd91
Merged revisions 7285,7299,7310,7329 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7285 | tilghman | 2005-12-02 15:12:05 -0600 (Fri, 02 Dec 2005) | 2 lines
Turn on executable bits for startup scripts, and fix bash var interpolation for Mandrake
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r7299 | oej | 2005-12-02 19:24:40 -0600 (Fri, 02 Dec 2005) | 2 lines
Documenting the default registerattempts setting as 0, continue hammering the server for ever and ever ;-)
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r7310 | tilghman | 2005-12-03 13:55:05 -0600 (Sat, 03 Dec 2005) | 3 lines
Bug 5925: check for "Unknown", as that's what app_voicemail puts into the field for Unknown callerid
Also, remove useless res checks (initialized to 0; never set)
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r7329 | kpfleming | 2005-12-04 12:03:07 -0600 (Sun, 04 Dec 2005) | 2 lines
use a more efficient way to get the revision number, that will also report if the working copy contains uncommitted modifications
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6a2bb9c3e0
add description of the fromdomain option (issue #5874 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
2c65582b66
remove extraneous svn:executable properties
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
754e28abaf
Cleanup language in sip.conf (bug #5543 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
aca0431935
properly document this morning's changes to 'ignoreregexpire'
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
de47e29d09
Merge highly modified fullcontact in realtime sip patch (bug #4832 , heavy mods)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
cc104b0066
finish commit bug 5427 (sip.conf.sample deserves to be updated too)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6745 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
aeaf9f4cc6
remove unused config option (issue #5358 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
f387421905
make sample config files easier to ready (issue #5371 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
dea1d768f9
update sample configuration to reflect new options (issue #5357 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8af7725f98
support optional sending of Remote-Party-ID headers (issue #2471 , heavily modified to actually work properly)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
fdccf66d52
add domains documentation and rename config option
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
d62b19048a
spelling fix (issue #5231 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
2f67f66577
Allow "auto" dtmf mode to select between RFC2833 and inband based on peer's offer or answer.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5f07eec58a
remove unused 'outgoinglimit' code, rename 'incominglimit' to 'call-limit' (old syntax is still supported) (issue #5068 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
bf6bc6d4fd
clean up rtcaching config options (issue #4075 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0bf1d48e40
support configurable Message-Account value in NOTIFY messages (issue #4801 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5736270107
add 'rtignoreexpire' option to allow Realtime addresses to be used regardless of expiration time (bug #4362 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
090f7d278b
more cleanups for outbound registrations (bug #4457 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
7e058413fb
allow fine-grained 'insecure' settings (bug #4024 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
03f5290b89
Add SIP real authentication (bug #3782 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
3fa3e8539d
Merge config sample updates mentioning reload (bug #3697 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
6a181832bb
Add callingpres/calling limit support, small updates (bug #3577 with mods)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5017 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
cecb25eae5
Add sip show user and friends (bug #3571 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
dd7e4ef47a
Merge SIP changes from oej
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
5d9bbe64c8
Merge tony's IAX/SIP realtime cache (bug #3509 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
8f199888d5
Merge callevents etc (bug #3456 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4921 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
cb06d1d954
Minor configuration fixes/standardizations (bug #3317 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
b705f19938
Fix small sip conf issues (bug #3296 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
7568d9a4ef
Merge OSP updates from matt nicholson (with changes)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
2f2783f231
Document polycom example (bug #3129 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
974ecf7830
Fix example to reflect that "never" is the default value for progressinband (bug #3129 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
b1b8802edd
Make it possible to configure an externhost and not just an externip (bug #3069 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
a2775a6109
fix documentation (bug #3127 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
1297800175
Make registertimeout be configurable (bug #2874 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
1a147cf804
Add user=phone option (bug #2244 , thanks oej)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
aaf49c6f4c
Add option for small headers (bug #2948 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
387fbae462
Update sample config files for setvar
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
732337d1e6
On second thought, add new "never" option for progressinband
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
3e1fa1f6ac
Fix sneakiness of OEJ
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
552b431691
Allow client code to be set from snom phone if "useclientcode=yes" in friend
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
879c401544
Add option to register extensions on IAX or SIP registration
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3746 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
f1be7d7ffc
Add new "route" mode to work around UNIDEN bugs (bug #2308 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
b27c9f26fe
Document caveats of promiscredir (bug #2195 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
21 years ago
deb02a000f
Add "username" to sip show peer (bug #2163 ) as well as a few config cleanups
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
ee4a248e6b
Make history configurable
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
ef0128ec41
Allow nat=never mode to work around buggy UNIDEN UIP200 firmware (bug #1935 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
9127ab071b
Add option for in-band progress (bug #1879 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
585afa7e8e
Add promiscuous redirect option
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
737e9b2458
Merge outgoing MSN support + Remote Party ID for SIP (bug #1841 ) with cleanups
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
dd4aba8022
Document ability to change user agent (bug #1867 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
3ef099dede
In spite of my extreme distaste for SRV, make it the default.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
e446f4ca81
Merge OSS fixes for FreeBSD, implement rtptimeout and rtpholdtimeout
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
9082b027ef
Improve sample configuration files (bug #1125 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
d2a4408930
Show peers formatting fixes, SIP strlen optimizations, and using ACL for local network matching (bugs #1587 , 1586, 1585, 1584)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2927 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
efc8c74b9a
Allow global authentication realm to be set (bug #1567 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
bbaf536f6a
Add "insecure=very" where we don't authenticate peers who are registered
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
7ca96ea7e5
Add white space condensing to pedantic mode of chan_sip.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
39c9375eef
Fix minor typo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
fb0af47761
Update sample sip.conf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
2414aeb4dd
Fix small typo in sip.conf.sample
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
b79b6aba5d
Bug # 1013: More explanation in the sip.conf.sample thanks to oej
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
057f297bfd
document localnet/proxy
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
0827fe8bc1
Don't do externip by default
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
da93b54687
Add externip keyword so that it's possible to use asterisk behind a NAT through port forwarding
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
cacfd42960
Add usecallingpres keyword to zapata.conf for PRI to PRI call so that we would take the presentation mode and send it to outbound call.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
22 years ago
eefb17607c
Add codec examples in sip.conf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
e4614cf487
Add optional pedantic SIP checking
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
f5e13431a5
Add SIP/RTP video support, video enable app_echo, start on RTCP
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
fc8d3631b4
Add SRV code to SIP, cleanup ENUM and make IAX2 do the right thing on dials
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
f7e921313e
Make mimetype for NOTIFY configurable
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
0a9130bb4b
SIP conf patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
c7c91a93f2
Allow "fromuser" option and don't append port number if it's specified (you can do it with fromdomain=temp.com:5060
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
43604eb770
Implement call pickup on SIP, override context if appropriate
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
060dda2022
Mon Mar 17 07:00:01 CET 2003
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
c1e327a713
Mon Mar 10 07:00:00 CET 2003
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
d50ea0e9d4
Sat Mar 8 07:00:00 CET 2003
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
9ea88c1825
Mon Feb 17 07:00:01 CET 2003
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
1c156b82d1
Sun Feb 16 07:00:01 CET 2003
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
32c3738fb8
Version 0.3.0 from FTP
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
23 years ago
ede7bca2a7
Version 0.1.12 from FTP
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
24 years ago
Mark Michelson
Olle Johansson
Dwayne M. Hubbard
Jason Parker
Joshua Colp
Luigi Rizzo
Russell Bryant
Kevin P. Fleming
Tilghman Lesher
Matthew Fredrickson
Mark Spencer
Josh Roberson
Malcolm Davenport
Jeremy McNamara
Martin Pycko
Matteo Brancaleoni