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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-19 20:44:39 +00:00
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
2006-07-13 20:35:41 +00:00
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-10 11:20:49 +00:00
b971f65978
- Make use of system name in realtime SIP peers optional
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- 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
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
2006-06-30 07:18:30 +00:00
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-29 08:01:08 +00:00
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
2006-06-29 07:04:43 +00:00
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
2006-06-26 18:34:29 +00:00
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
2006-06-26 16:24:43 +00:00
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
2006-06-13 19:38:41 +00:00
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
2006-06-01 16:47:28 +00:00
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
2006-06-01 12:43:01 +00:00
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
2006-05-31 16:56:50 +00:00
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
2006-05-24 03:28:49 +00:00
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
2006-05-18 16:57:59 +00:00
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
2006-05-18 14:07:46 +00:00
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
2006-05-11 13:54:00 +00:00
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
2006-05-08 15:46:02 +00:00
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
2006-05-02 20:31:39 +00:00
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
2006-04-28 16:42:42 +00:00
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
2006-04-23 06:22:29 +00:00
023e27f695
Formatting fixes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-06 15:23:14 +00:00
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
2006-04-04 08:01:46 +00:00
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
2006-03-28 04:21:21 +00:00
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
2006-03-28 03:28:52 +00:00
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
2006-03-27 03:35:49 +00:00
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
2006-03-27 02:57:17 +00:00
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
2006-03-19 09:35:11 +00:00
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
2006-03-16 18:01:08 +00:00
6b8701cffa
Whitespace changes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-28 21:04:17 +00:00
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
2006-02-10 16:33:47 +00:00
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
2006-02-09 15:40:53 +00:00
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
2006-02-01 13:23:59 +00:00
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
2006-01-30 19:50:39 +00:00
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
2006-01-25 12:01:07 +00:00
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
2006-01-24 18:15:20 +00:00
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
2006-01-20 23:19:49 +00:00
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
2006-01-20 14:32:30 +00:00
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
2006-01-04 09:10:56 +00:00
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
2006-01-03 11:21:48 +00:00
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
2005-12-22 20:38:43 +00:00
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
2005-12-04 18:12:52 +00:00
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
2005-11-30 05:26:29 +00:00
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
2005-11-29 18:24:39 +00:00
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
2005-10-29 21:05:34 +00:00
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
2005-10-28 16:00:09 +00:00
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
2005-10-28 13:37:31 +00:00
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
2005-10-12 01:46:42 +00:00
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
2005-10-05 00:05:13 +00:00
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
2005-10-04 22:51:59 +00:00
Kevin P. Fleming
Olle Johansson
Joshua Colp
Russell Bryant
Matthew Fredrickson
Mark Spencer
Josh Roberson