kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-11-14 05:48:54 +00:00
Code Issues Packages Projects Releases Wiki Activity
13,220 Commits 151 Branches 1,613 Tags
cefe4f025d1f23bc11ca901b583f0d4fc562f6b6
Commit Graph

156 Commits

Author SHA1 Message Date
Olle Johansson
566429c300 Add a better explanation of the difference between the device namespace and the dialplan for newbies. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@171837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2009-01-28 13:07:27 +00:00
Tilghman Lesher
a4ebc105ef Create rules for disallowing contacts at certain addresses, which may 
improve the security of various installations.  As this does not change
any default behavior, it is not classified as a direct security fix for
anything within Asterisk, but may help PBX admins better secure their
SIP servers.
(closes issue #11776)
 Reported by: ibc
 Patches: 
       20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76, blitzrage


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@142865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-09-12 20:37:18 +00:00
Tilghman Lesher
fc195a2df6 More fixes for realtime peers. 
(closes issue #12921)
 Reported by: Nuitari
 Patches: 
       20080804__bug12921.diff.txt uploaded by Corydon76 (license 14)
       20080815__bug12921.diff.txt uploaded by Corydon76 (license 14)
 Tested by: Corydon76


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@138258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-08-15 22:33:42 +00:00
Russell Bryant
960ebde9ca Comments in this config file were aligned only if your tab size was set to 8. 
So, convert tabs to spaces so that things should be aligned regardless of what
tab size you use in your editor.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@137731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-08-14 14:05:23 +00:00
Olle Johansson
fcb5675ffb Clear up documentation on "domain=" setting in sip.conf 
Reported by: davidw
(closes issue #12413)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@126844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-07-01 12:53:01 +00:00
Tilghman Lesher
3841c847a3 Correct description of notifyringing option. 
(Closes issue #12890)
Reported by gminet


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@123883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-06-19 16:20:41 +00:00
Joshua Colp
405dfcb54a Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. 
(closes issue #10417)
Reported by: cstadlmann


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-05-28 14:23:34 +00:00
Tilghman Lesher
6d2c05cbec Reference documentation files that actually exist. 
(closes issue #12516)
 Reported by: linuxmaniac
 Patches: 
       diff_rev114611.patch uploaded by linuxmaniac (license 472)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2008-04-25 15:53:52 +00:00
Olle Johansson
8b650ee007 Clarify limitonpeers=yes 
(closes issue #11304)
Reported by: pj



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-11-27 07:34:19 +00:00
Jason Parker
070bcf111e Correct the allowexternaldomains option in SIP sample config. 
Issue 10753


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@82751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-09-18 15:28:21 +00:00
Joshua Colp
c98e199fb2 (closes issue #10335) 
Reported by: adamgundy
Update sip.conf to include another scenario where directrtpsetup will fail.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-08-08 13:51:01 +00:00
Joshua Colp
fa866efb5c 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@58779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-03-12 00:51:16 +00:00
Olle Johansson
90a4b844a9 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@53109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-02 00:24:03 +00:00
Olle Johansson
d7cde47f06 Add explanation of port= in combination with defaultip= (thanks jsmith) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-01 16:35:12 +00:00
Olle Johansson
d2b7e8b247 Be a bit more politically correct 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 17:59:53 +00:00
Olle Johansson
bfe4bb0f1e Issue #8575 - Buggy cisco MWI support. 
Normally we try not to change our software for bugs in other devices. But in
this case, the Cisco phones are so widespread so we try to implement a fix while
waiting for a bugfix from Cisco.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-27 16:49:45 +00:00
Olle Johansson
7945d4ca35 Add missing s from another repository. (thanks jcmoore!) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-05 15:59:05 +00:00
Olle Johansson
027096b3a3 Updating sip.conf.sample with information about T38 not working 
when chan_local or chan_agent is involved in the call.

I don't know how big a fix that would be to solve, but this is
the current state of affairs.

(Chan_sip currently checks if the other side of the bridge
has a SIP tech. We could/should implement another check,
possibly for udptl_write or some flag in the ast_channel
structure).


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-05 12:39:30 +00:00
Olle Johansson
f89143bd13 - Disable RTP hold timers while T.38 fax transmission happens 
- Encapsulate RTP timers in the rtp structure so we have one for video and one for audio
   The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send
   something that video phones support in the RTP stream.
   I now this is a big architectual change at this stage for 1.4, but decided it was needed
   to avoid future bug reports.
- Document the RTP NAT keepalive option in sip.conf.sample

Issue 7679 in the bug tracker. Please test.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-02 11:32:51 +00:00
Olle Johansson
98d3fb64ed - Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states 
- Remove support for T.38 early media, since it's impossible.

(Two patches in one - extra friday evening offer due to being off line from svn today... :-)


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-12-01 17:41:56 +00:00
Joshua Colp
802c3c3ecf Merged revisions 48142 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
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)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-30 17:57:35 +00:00
Olle Johansson
a68edf400f Explain the use device status system implemented in SIP for subscriptions, 
queues and manager a bit better.

Like in 1.2, you will get more detailed information if you set a call 
limit for a device. When the call limit is reached, the status system will
report a device as busy.

For queues, setting a call limit per SIP device is propably a requirement.

In most cases, it will work much better if you only use type=peer and not
type=friend. We might decide to backport the new setting from trunk to
apply all call limits to the peer part of a friend only.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-29 19:56:56 +00:00
Olle Johansson
3fe8e34039 Clarify RTP timers. Sorry, grandma. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-29 08:03:36 +00:00
Olle Johansson
7da1a54fe6 Explain properly how videosupport works. 
Committ from Asterisk Video Task Force meeting in Paris!


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-20 11:45:22 +00:00
Olle Johansson
e1e6a1b2a8 Make the HOLD notification optional, in order to avoid a lot of extra database lookups 
for all those realtime users out there.



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-16 19:24:23 +00:00
Olle Johansson
5bd53e3588 - CANCEL is never authenticated (according to the RFC) 
- Update docs on canreinvite. "nonat" is the recommended setting for most users with
  phones behind a NAT.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-11-16 15:03:49 +00:00
Olle Johansson
9ab1cc22a4 Support ;rport when we're supposed to support ;rport. Issue #7473. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-31 10:26:16 +00:00
Olle Johansson
590698e583 Adding information about Marks direct-RTP hack to the docs... 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-17 17:39:18 +00:00
Olle Johansson
45fc0eaba4 Now, remove all traces of the option that we did not need :-) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-17 16:23:27 +00:00
Joshua Colp
d28fd24747 Merged revisions 45265 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
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.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-16 20:06:18 +00:00
Joshua Colp
3f24dceeca Merged revisions 45260 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
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.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-16 19:37:34 +00:00
Jason Parker
8bd82ebc0d Add documentation on rtp packetization. 
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
 2006-09-20 17:39:59 +00:00
Tilghman Lesher
091e1aed8d Merged revisions 42716 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r42716 | tilghman | 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) | 2 lines

Spelling/grammar fixes (Issue 7929)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-09-11 16:41:49 +00:00
Joshua Colp
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! 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-08-31 01:59:02 +00:00
Kevin P. Fleming
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) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-07-19 20:44:39 +00:00
Kevin P. Fleming
4376af0080 actually make the non-standard G726-32 behavior available for SIP clients 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-07-13 20:35:41 +00:00
Olle Johansson
0e0059c0f3 Remove configuration option "restrictcid" that is nowhere to 
be seen in the code. Did it exist, was it planned to exist
or was it documentationware only? Ask Dr Asterisk.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-07-10 11:20:49 +00:00
Olle Johansson
b971f65978 - Make use of system name in realtime SIP peers optional 
- Fix small issue with SIP history


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-07-02 12:00:36 +00:00
Olle Johansson
f3594bd1a0 Removing configuration options that does not do anything yet. No need to 
add "promises" to the sip.conf.sample...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-30 07:18:30 +00:00
Kevin P. Fleming
dec3d7d4c0 Merged revisions 36253-36254 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r36253 | kpfleming | 2006-06-29 02:19:27 -0500 (Thu, 29 Jun 2006) | 2 lines

add documentation for peer-specific 'outboundproxy' setting

........
r36254 | kpfleming | 2006-06-29 02:19:54 -0500 (Thu, 29 Jun 2006) | 2 lines

clarify documentation for 'persistentmembers' setting

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-29 08:01:08 +00:00
Olle Johansson
4177596e8d reformatting sip.conf.sample a bit, adding dumphistory that was not documented 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-29 07:04:43 +00:00
Olle Johansson
cc43f0bdc7 Speling error. Avoid swenglish :-) (thanks, jtodd!) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-26 18:34:29 +00:00
Olle Johansson
e2b0c5b558 Add example of permit/deny to sip.conf.sample 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-26 16:24:43 +00:00
Joshua Colp
5456f425c6 Allow AST_FRAME_MODEM frames to be dumped, and document T.38 passthrough support 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-13 19:38:41 +00:00
Russell Bryant
4c76028de9 - add the ability to configure forced jitterbuffers on h323, jingle, 
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
 2006-06-01 16:47:28 +00:00
Kevin P. Fleming
6bce269454 Merged revisions 31321 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
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)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-01 12:43:01 +00:00
Russell Bryant
bb7dd96cfe Add support for using a jitterbuffer for RTP on bridged calls. This includes 
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
 2006-05-31 16:56:50 +00:00
Kevin P. Fleming
3e99be68d1 add a new option for 'obscuring' SIP user/peer names from fishers 
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
 2006-05-24 03:28:49 +00:00
Kevin P. Fleming
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) 
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
 2006-05-18 16:57:59 +00:00
Joshua Colp
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) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-18 14:07:46 +00:00