kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-31 02:37:10 +00:00
Code Issues Packages Projects Releases Wiki Activity
10,017 Commits 151 Branches 1,609 Tags
b58cc9e1bd3c545e270ad7d8d39bf26c7d8c0ae7
Commit Graph

131 Commits

Author SHA1 Message Date
Joshua Colp
b58cc9e1bd Merged revisions 45262 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
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

........
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/trunk@45263 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-16 19:43:33 +00:00
Olle Johansson
77c69dc4ef Recommend using "sip reload" since it's much easier to learn and 
remember.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-07 16:26:11 +00:00
Luigi Rizzo
b19b4b9764 document a bit the use of templates. 
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: https://origsvn.digium.com/svn/asterisk/trunk@44579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-06 16:43:36 +00:00
Luigi Rizzo
f94849ca2a document the "contact" option a bit better. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-06 16:20:42 +00:00
Luigi Rizzo
ccca5843fd Two things: 
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: https://origsvn.digium.com/svn/asterisk/trunk@44566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-06 15:41:12 +00:00
Luigi Rizzo
2a7ac3f735 update example commands to match current syntax 
(does not apply to 1.4)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-10-06 06:43:49 +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
Olle Johansson
5237a0e06d - Use systemname for realm in sip, if we have no configuration for realm 
- 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
Olle Johansson
ca6cf552f9 Add documentation on "allowtransfer" 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-08 15:46:02 +00:00
Olle Johansson
7bbb6bd3aa - fix typo in rtp.c, devicestate.h 
- 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
Russell Bryant
c38fbd246e note that group assignments must be from 0 to 63 (issue #7048) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-28 16:42:42 +00:00
Olle Johansson
5873462c2e - Add doxygen documentation for sipsock_read locking 
- 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
Olle Johansson
023e27f695 Formatting fixes 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-06 15:23:14 +00:00
Olle Johansson
95de51526a Added information on call-limit and realtime 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-04 08:01:46 +00:00
Kevin P. Fleming
8410e0d681 support subscription-based MWI, and use proper Call-ID on NOTIFY messages (issue #6390) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-28 04:21:21 +00:00
Kevin P. Fleming
278b8e8fc7 improve IP TOS support for SIP and IAX2 (issue #6355, code from jcollie plus modifications) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-28 03:28:52 +00:00
Olle Johansson
83d9331261 Issue #5427 
- 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
Olle Johansson
18de2b7787 Issue #6705 (oej) 
- 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
Olle Johansson
d7b5a18f4c Fix reference to README files 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-19 09:35:11 +00:00
Olle Johansson
1a206c1bf8 Clarify documentation for "progressinband" - imported from 1.2 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-03-16 18:01:08 +00:00
Olle Johansson
6b8701cffa Whitespace changes 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-02-28 21:04:17 +00:00
Kevin P. Fleming
b40bd71a9a restore 'rfc2833' naming for DTMF mode in chan_sip 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-02-10 16:33:47 +00:00
Olle Johansson
4d07b89fdd - Change "rfc2833" to "rtp" in sip.conf. Keeping backwards compatibility. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-02-09 15:40:53 +00:00
Olle Johansson
3f6cc5c544 - Clarify default setting of canreinvite (thanks royk) 
- 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
Olle Johansson
125fd8446c Issue 5892: Set a minimum T1 timer for calls. Reporter: twisted 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-01-30 19:50:39 +00:00
Olle Johansson
b64404e039 From now on, apply maxexpiry and minexpiry to all subscriptions. Thanks to fourcheeze in the IRC channel 
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
Olle Johansson
0ba27e0a6b Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug #6183) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-01-24 18:15:20 +00:00
Matthew Fredrickson
4dc76fbcc1 Fix comments in sip.conf (#6134) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-01-20 23:19:49 +00:00
Olle Johansson
125db028c3 - Add DOC file about caller ID presentation values 
- 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
Olle Johansson
5462ec082c - Remove "incominglimit" as a configuration option in sip.conf 
- 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
Olle Johansson
3b4f660a85 Bug 5345; Add configuration option for minimum registration time. (folsson) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-01-03 11:21:48 +00:00
Russell Bryant
b60daeb58f Merged revisions 7599 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
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

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2005-12-22 20:38:43 +00:00