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r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines
Improve T.38 negotiation by exchanging session parameters between application and channel.
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r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines
This patch adds a new 'ignoresdpversion' option to sip.conf. When this is
enabled (either globally or for a specific peer), chan_sip will treat any SDP
data it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session version
received is different from the current SDP session version. This option is
required to interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with Microsoft OCS
which always uses 0 as the session version).
http://reviewboard.digium.com/r/94/
(closes issue #13958)
Reported by: toc
Tested by: toc
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r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines
Accept T.38 re-INVITE responses with invalid SDP versions.
This commit changes the 'incoming SDP version' check logic a bit more; when
'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to
switch to T.38, we'll always accept the peer's SDP response, even if they
don't properly increment the SDP version number as they should. If this situation
occurs, a warning message will be generated suggesting that the peer's
configuration be changed to include the 'ignoresdpversion' configuration option
(although ideally they'd fix their SIP implementation to be RFC compliant).
AST-221
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r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
Correct documentation for the register line, specifically where the domain should be specified.
(closes issue #14367)
Reported by: Nick_Lewis
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This change does not affect any other 1.6 branches as they have
already been updated for other changes, which uses the word 'domain'
as I have here.
(closes issue #15204)
Reported by: okrief
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r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines
SIP set outbound transport type from Registration
In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset.
(closes issue #12282)
Reported by: rjain
Patches:
reg_patch_1.diff uploaded by dvossel (license 671)
Tested by: dvossel
(closes issue #14727)
Reported by: pj
Patches:
reg_patch_3.diff uploaded by dvossel (license 671)
Tested by: pj, dvossel
Review: https://reviewboard.asterisk.org/r/249/
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r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines
Merged revisions 186059 via svnmerge from
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r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines
Merged revisions 186056 via svnmerge from
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r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines
Fix for AST-2009-003
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Provide correct hint to debug SIP trouble in the default config
(closes issue #14646)
Reported by: strk
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r171838 | oej | 2009-01-28 14:11:44 +0100 (Ons, 28 Jan 2009) | 10 lines
Merged revisions 171837 via svnmerge from
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r171837 | oej | 2009-01-28 14:07:27 +0100 (Ons, 28 Jan 2009) | 2 lines
Add a better explanation of the difference between the device namespace and the dialplan for newbies.
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r168711 | oej | 2009-01-15 18:55:53 +0100 (Tor, 15 Jan 2009) | 4 lines
Clarify some misunderstandings and make it even more clear that you can refer to a peer
in the register= line.
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r142866 | tilghman | 2008-09-12 15:49:46 -0500 (Fri, 12 Sep 2008) | 18 lines
Merged revisions 142865 via svnmerge from
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r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines
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
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r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) | 16 lines
Merged revisions 138258 via svnmerge from
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r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines
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
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r137732 | russell | 2008-08-14 09:15:50 -0500 (Thu, 14 Aug 2008) | 12 lines
Merged revisions 137731 via svnmerge from
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r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines
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.
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r135126 | tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines
SIP should use the transport type set in the Moved Temporarily for the next
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
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r128524 | oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines
- Fixing issues with "sip show settings"
- 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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r128237 | oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines
Make TCP disabled by default (it's considered experimental)
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r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2 lines
Reformatting the config sample
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r127154 | bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines
Add a configuration option so the global outboundproxy can use tcptls without it being defined by each sip user.
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r126845 | oej | 2008-07-01 14:54:57 +0200 (Tis, 01 Jul 2008) | 14 lines
Merged revisions 126844 via svnmerge from
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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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r125891 | bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines
Change the way that the transport option works for sip users. transport will 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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r123887 | tilghman | 2008-06-19 11:21:32 -0500 (Thu, 19 Jun 2008) | 12 lines
Merged revisions 123883 via svnmerge from
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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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r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines
Merged revisions 118646 via svnmerge from
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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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r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines
Note that the TCP and TLS support is currently considered experimental and
is subject to change while we work out the remaining issues.
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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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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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(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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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));
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76221 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68662 f38db490-d61c-443f-a65b-d21fe96a405b
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else seemed to think was very funny. Oh well ... :)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67895 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66856 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66818 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@66777 f38db490-d61c-443f-a65b-d21fe96a405b
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