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invite.
(closes issue #11843)
Reported by: pestermann
Patches:
20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36)
20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36)
Tested by: pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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driver into a common place for multiple channel drivers.
(closes issue #13152)
Reported by: caio1982
Patches:
atxfer_complete_sound3.diff uploaded by caio1982 (license 22)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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fail to setup video RTP if the two endpoints will not support it. This assists
with call files and certain transfers to ensure that if two video phones are
ever connected, they will always share a video feed.
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and "tlsbindaddr".
Note: I don't think we can start properly without UDP port open, that needs to be tested.
- Removing "bindport" from configuration example, not needed to mention this any more
I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1)
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- Adding IP address for TCP and/or TLS too if auto-domain is enabled and
binding to a different IP address
- Fixing documentation in sip.conf.sample
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without it being defined by each sip user.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines
Clear up documentation on "domain=" setting in sip.conf
Reported by: davidw
(closes issue #12413)
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now take multiple arguments, the first one listed will be the one used
for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on
the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason.
(issue #12799)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines
Correct description of notifyringing option.
(Closes issue #12890)
Reported by gminet
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines
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
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basis in the register line. This comes from a Switchvox patch. (issue AST-24)
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Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
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astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it.
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to be played upon completion of an attended transfer.
(closes issue #9239)
Reported by: sunder
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is subject to change while we work out the remaining issues.
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(closes issue #11959)
Reported by: mostyn
Patches:
peerstatus3.patch uploaded by mostyn (license 398)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11587)
Reported by: sobomax
Patches:
dialstring_doc.diff uploaded by sobomax (license 359)
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valid ciphers provided in both files. .. First commit since July, woot
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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
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branch.
This set of changes introduces SIP session timers support (RFC 4028). In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.
To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."
(closes issue #10665)
Reported by: rjain
Patches:
chan_sip.c.1.diff uploaded by rjain (license 226)
chan_sip.c.diff uploaded by rjain (license 226)
sip.conf.sample.diff uploaded by rjain (license 226)
proc_422_rsp_comment.diff uploaded by rjain (license 226)
chan_sip.c.cache.diff uploaded by rjain (license 226)
chan_sip.memalloc uploaded by rjain (license 226)
chan_sip.memalloc.bugfix uploaded by rjain (license 226)
Patches tracked in team/group/sip_session_timers, with some additional fixes
by russell and oej.
Tested by: jtodd, rjain, loloski
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to set the qualify frequency.
(closes issue #11597)
Reported by: wilder
Patches:
qualifyfreq5.patch uploaded by wilder (license 362)
-- with some mods by me
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by adding an exclamation mark to the dial string.
This patch also exists for 1.4 in the fixtoheader-1.4 branch
and has been in production for quite some time.
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- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
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non-responsive SIP servers.
Thanks, jcmoore, for the patch!
Reported by: jcmoore
Patches:
peer_t1_timerb_trunk_v3.patch.txt uploaded by jcmoore (license 9)
(closes issue #9771)
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"Username" still works, but is deprecated.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines
Clarify limitonpeers=yes
(closes issue #11304)
Reported by: pj
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Both still works in this version.
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call-limits
and we now have the groupcount system to implement call-limits in the dialplan. You
can use the "setvar" option in realtime/sip.conf to set limits per device.
- Implement "callcounter" as a new option to enable the call counting we need to
report device status to queue, manager and SIP subscriptions.
The call counter setting is now enabled in the code by setting the device call-limit
to 999. When we remove the call limit, we can simply enable this with a boolean
setting.
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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)
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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
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session field, closes issue# 10795
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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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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.
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Reported by: IgorG
Patches:
sip_conf-80933-1.patch uploaded by IgorG (license 20)
Fix up sip.conf sample configuration.
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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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+ 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));
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else seemed to think was very funny. Oh well ... :)
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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.
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becomes unreachable
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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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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.
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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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