10d1b9347c
Use ast_str_append() instead of ast_build_string() to construct
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the sdp messages. Overall the code is slightly more readable
(because the string is fully described by a single pointer),
and more efficient (because the length is stored explicitly
so you don't need to do strlen()).
(I have been using this code for almost a year now.)
I wish we had infix string operators to do this sort of things!
Nothing to backport from this change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-21 02:33:25 +00:00
06a3436375
We have two 'technology' descriptors for a SIP channel, so
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define and use a macro to determine whether we are pointing to
one of them, so when one goes away (or a new one appears) we don't
have to touch all the code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-21 01:25:13 +00:00
2286afa3af
Enhance NAT support as discussed on the -dev list, i.e.:
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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: https://origsvn.digium.com/svn/asterisk/trunk@76221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-21 01:01:10 +00:00
989b93143a
Merged revisions 76087 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76087 | file | 2007-07-20 14:20:09 -0300 (Fri, 20 Jul 2007) | 14 lines
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r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6 lines
(closes issue #10247 )
Reported by: fkasumovic
Patches:
chan_sip.patch uploaded by fkasumovic (license #101 )
Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer.
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2007-07-20 17:21:23 +00:00
66cae9269b
It is impossible for the externhost variable to not exist, it is however possible for it to be empty.
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2007-07-20 16:51:09 +00:00
bfc782f4e9
Don't use a field size for the last argument of printf format,
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because in this case the string is left-aligned and it is not
truncated anyways.
Omitting the field size prevents the generation of trailing whitespace,
which makes the string fit in smaller windows.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-20 15:06:54 +00:00
b2fec9ad16
Extend the 'network settings' section with indication on the
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localnet settings (requires the change in SVN 76034), and also
give an indication on whether/why/how the remapping of addresses
in SIP message is done or not.
I think this is especially useful for debugging the configuration,
as the address remapping depends on a combination of at least 3
parameters (localnet, externhost, externip) and successful DNS lookup.
An example of the output of this section is below:
Network Settings:
---------------------------
SIP address remapping: Enabled using externhost
Externhost: foo.dyndns.net
Externip: 80.64.128.23:0
Externrefresh: 10
Internal IP: 12.34.56.78:5060
Localnet: 192.168.0.0/255.255.0.0
10.0.0.0/255.0.0.0
I leave to the community the judgement if the above info is a
useful addition for 1.4. It is not a bugfix, but it is neither a
new feature, only a useful diagnostic tool.
Note that I would like to move there also the bindaddress/port
information, in the usual addr:port format e.g.
Bindaddress: 0.0.0.0:5060
so that network information is all in one place.
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2007-07-20 14:54:01 +00:00
0e969271ae
After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
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2007-07-19 23:24:27 +00:00
d60c5ee296
print more of the network settings (externip, externhost etc.)
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in the "sip show settings" cli output. I have put these in a
separate section, probably even bindaddr and SIP port should go
there.
There are more things to add here e.g. localnet and so on.
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2007-07-19 08:07:04 +00:00
192ac53c3f
document the use of externip, externhost and other nat-related options,
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as well as the handling of the sip socket.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-19 08:00:03 +00:00
fddd5b557c
ast_sip_ouraddrfor() never fails, so make it void
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and remove the code that would never be called.
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2007-07-19 07:51:34 +00:00
00d9a3e7a0
portability fix: use %f instead of %lf when printing double.
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The l is useless.
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2007-07-19 07:41:45 +00:00
81bc1d7af5
Merge in ast_strftime branch, which changes timestamps to be accurate to the microsecond, instead of only to the second
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2007-07-18 19:47:20 +00:00
a23feea9d2
Merged revisions 75623 via svnmerge from
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r75623 | file | 2007-07-18 12:44:02 -0300 (Wed, 18 Jul 2007) | 2 lines
Few more places that needs to check for onhold state.
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2007-07-18 15:45:18 +00:00
d90bddfa6c
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r75621 | file | 2007-07-18 12:41:06 -0300 (Wed, 18 Jul 2007) | 5 lines
(closes issue #10165 )
Reported by: elandivar
It is possible for hold status to exist without call limits set, so we need to ensure update_call_counter is executed regardless.
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2007-07-18 15:42:11 +00:00
5ac24b25d3
This corrects the problem with flags and %lld formats on 64-bit machines, where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc.
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2007-07-18 14:35:07 +00:00
4003b31fc5
Minor code tweaks. Variables were being checked wrong in some situations and didn't need to be checked in others.
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2007-07-18 13:59:01 +00:00
8a7732f067
via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
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2007-07-17 19:40:29 +00:00
6bc0a4929c
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r74955 | murf | 2007-07-12 14:42:08 -0600 (Thu, 12 Jul 2007) | 1 line
This patch resolves 10143; thanks to irroot for the patch; looked acceptable. Let the community decide if it messes things up
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2007-07-12 20:46:32 +00:00
a1b9cbcd31
Implementation of a feature that will disable "missed calls" counters on SIP phones.
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If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.
Thanks to Ramon and Frank for feedback on this feature.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-09 08:27:37 +00:00
ba857cc8a9
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r73985 | tilghman | 2007-07-08 23:03:20 -0500 (Sun, 08 Jul 2007) | 2 lines
Doxygen formatting fixes; fixes errors while 'make progdocs'. (Closes issue #10104 )
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2007-07-09 04:09:16 +00:00
74e8ab14fc
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r73849 | oej | 2007-07-08 11:47:31 +0200 (Sun, 08 Jul 2007) | 2 lines
While tracking down a bug, I need some more history. Dumphistory is very useful, indeed.
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2007-07-08 09:49:21 +00:00
1da115c8d9
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r73769 | russell | 2007-07-06 18:02:58 -0500 (Fri, 06 Jul 2007) | 12 lines
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r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | 4 lines
If a sip_pvt struct has already registered an extension state callback,
remove the old one before adding a new one. If this isn't done, Asterisk
will crash. (issue #10120 )
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2007-07-06 23:05:24 +00:00
a0c37d2548
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r73679 | russell | 2007-07-06 10:57:25 -0500 (Fri, 06 Jul 2007) | 15 lines
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r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | 7 lines
(closes issue #10125 )
Reported by: makoto
Patches submitted by: makoto
This fixes a crash in chan_sip that happens when the bindaddr setting is not
valid on Asterisk startup, gets fixed, and then a reload gets issued.
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2007-07-06 16:00:03 +00:00
134a556c9f
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r73598 | russell | 2007-07-05 18:59:22 -0500 (Thu, 05 Jul 2007) | 3 lines
Fix a crash in chan_sip. Don't try to stop the monitor thread if it was never
started. (closes issue #10124 , reported by gzero, fixed by me)
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2007-07-05 23:59:50 +00:00
cc19ba80f5
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r73548 | kpfleming | 2007-07-05 17:20:44 -0500 (Thu, 05 Jul 2007) | 10 lines
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r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) | 2 lines
we shouldn't allow G.723.1 endpoints to use VAD, just like we don't support it for G.729
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2007-07-05 22:29:37 +00:00
0fc25ac3ee
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r73467 | file | 2007-07-05 16:18:02 -0300 (Thu, 05 Jul 2007) | 10 lines
Merged revisions 73466 via svnmerge from
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r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 lines
Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique)
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2007-07-05 19:20:12 +00:00
daec10d187
Fix building with -Wdeclaration-after-statement, here too
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2007-06-28 19:35:23 +00:00
62084eb2a4
Add SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables when a transfer takes place. (issue #8378 reported by jcovert)
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2007-06-27 23:13:09 +00:00
1961b57705
Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to.
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2007-06-26 23:31:23 +00:00
d77301b8cd
Tweak CLI command completion and some help text. (issue #10049 reported by IgorG)
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2007-06-25 15:35:10 +00:00
76455dda03
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r71430 | file | 2007-06-24 21:10:06 -0400 (Sun, 24 Jun 2007) | 10 lines
Merged revisions 71414 via svnmerge from
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r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 lines
Ignore other URIs after the first in a 300 Multiple Choice response. (issue #10041 reported by homesick)
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2007-06-25 01:11:47 +00:00
18f4920227
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r70552 | file | 2007-06-20 18:22:20 -0400 (Wed, 20 Jun 2007) | 10 lines
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r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 lines
Don't overwrite the configured username setting upon a REGISTER. (issue #8565 reported by jsmith)
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2007-06-20 22:24:47 +00:00
238b7a54cc
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r69944 | russell | 2007-06-19 10:22:36 -0500 (Tue, 19 Jun 2007) | 10 lines
Fix a crash that could occur when handing device state changes.
When the state of a device changes, the device state thread tells the extension
state handling code that it changed. Then, the extension state code calls the
callback in chan_sip so that it can update subscriptions to that extension.
A pointer to a sip_pvt structure is passed to this function as the call which
needs a NOTIFY sent. However, there was no locking done to ensure that the pvt
struct didn't disappear during this process.
(issue #9946 , reported by tdonahue, patch by me, patch updated to trunk to use
the sip_pvt lock wrappers by eliel)
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2007-06-19 15:27:16 +00:00
a67890d7a9
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r69796 | tilghman | 2007-06-18 14:48:17 -0500 (Mon, 18 Jun 2007) | 2 lines
Issue 10005 - Segfault with missing arguments, plus fix a missing define for SIP INFO channels
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2007-06-18 19:52:56 +00:00
9ed0563f17
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r69794 | file | 2007-06-18 15:00:50 -0400 (Mon, 18 Jun 2007) | 2 lines
Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc)
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2007-06-18 19:02:45 +00:00
59bc48bd05
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r69775 | file | 2007-06-18 14:18:12 -0400 (Mon, 18 Jun 2007) | 10 lines
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r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 lines
Set the peer name on the dialog to the one configured in sip.conf and NOT the username to be used for authentication attempts. (issue #9967 reported by achauvin)
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2007-06-18 18:19:54 +00:00
1dbfbe6d71
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r69668 | file | 2007-06-18 12:04:55 -0400 (Mon, 18 Jun 2007) | 2 lines
Don't defer the BYE till later on a transfer when the transfer itself goes kaboom and has no hope of working.
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2007-06-18 16:06:17 +00:00
3e4980da79
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r69661 | file | 2007-06-18 11:46:32 -0400 (Mon, 18 Jun 2007) | 2 lines
Few minor transfer tweaks. We can't unlock something we never locked, and better handle a specific scenario with doing an attended transfer between two non-bridged calls.
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2007-06-18 15:48:05 +00:00
981a94f023
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r69625 | file | 2007-06-18 09:55:00 -0400 (Mon, 18 Jun 2007) | 2 lines
Fix issue where it would be possible for the negotiated codecs to get set back to nothing. (issue #9992 reported by yehavi)
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2007-06-18 13:57:33 +00:00
055d82cbce
Add a massive set of changes for converting to use the ast_debug() macro.
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(issue #9957 , patches from mvanbaak, caio1982, critch, and dimas)
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2007-06-14 19:39:12 +00:00
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r69183 | russell | 2007-06-13 14:57:38 -0500 (Wed, 13 Jun 2007) | 9 lines
Move the logic for destroying a call when no response is received to a BYE
outside of the block that checks for FLAG_FATAL to be set. This flag is only
set when the packet is transmitted with the reliability set to XMIT_CRITICAL
when the original packet is transmitted. A BYE is always sent with it set
to XMIT_RELIABLE, meaning this code could never be encountered. This resulted
in seeing some SIP channels that would never go away with the last packet
sent being a BYE.
(part of issue #9235 , patch from jcmoore)
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2007-06-13 20:03:03 +00:00
63535ada60
Fixes for ast_strlen_zero() janitor project.
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Issue 9968, patch by eliel.
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2007-06-13 17:06:53 +00:00
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r69071 | russell | 2007-06-13 11:56:16 -0500 (Wed, 13 Jun 2007) | 3 lines
Clarify a bit of logic. This doesn't change behavior in any way, but it is
helpful when following the logic to debug problems like 9235.
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2007-06-13 16:59:42 +00:00
1d57ccb6f7
Fix a bunch of doxygen errors and document more things
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(issue #9842 , snuffy)
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2007-06-07 23:07:25 +00:00
1f9d98016e
- Doxygen updates
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- Adding docs on flags to be able to clean up a bit
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2007-06-07 19:45:32 +00:00
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r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 lines
Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia)
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2007-06-07 00:12:21 +00:00
6aec360466
Remove our little joke that was making fun of email disclaimers which nobody
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else seemed to think was very funny. Oh well ... :)
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2007-06-06 22:27:18 +00:00
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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2007-06-06 21:20:11 +00:00
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r67862 | russell | 2007-06-06 16:14:46 -0500 (Wed, 06 Jun 2007) | 4 lines
Fix a crash when doing call pickups with SIP phones. The code unlocked the
channel when it should not have.
(issue #9652 , reported by corruptor, fixed by me)
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2007-06-06 21:16:18 +00:00
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r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2 lines
Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo)
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2007-06-04 19:32:08 +00:00
93f2be2675
Fix a couple of places where "tos" was used instead of "cos".
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(issue #9540 , patch by IgorG)
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2007-06-04 15:14:52 +00:00
3d2b58751f
To satisfy some legal concerns, add an option for chan_sip to include a
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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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r66768 | file | 2007-05-31 12:14:48 -0400 (Thu, 31 May 2007) | 10 lines
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r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines
It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx)
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2007-05-31 16:18:14 +00:00
0b2db74e5a
Issue #9842 - Doxygen updates by snuffy. Thanks!
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(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference)
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2007-05-31 10:26:55 +00:00
4c1068c136
oops. Thanks Terry.
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2007-05-29 19:53:40 +00:00
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r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2 lines
Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response.
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2007-05-29 19:35:43 +00:00
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r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2 lines
Don't issue hangup on hangup on hangup on hangup (for jcmoore)
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2007-05-29 19:17:49 +00:00
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r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2 lines
Don't reset hangupcause if we already have one
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2007-05-29 16:19:53 +00:00
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r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2 lines
Tracking down hanging channels, killing them one by one. Issue #9235 and related
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2007-05-29 16:17:17 +00:00
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r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, 29 May 2007) | 10 lines
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r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines
Issue #9802 - Change inuse counter on CANCEL
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2007-05-29 10:02:31 +00:00
4d03b4f268
Don't try to unregister a peer using the sip unregister CLI command if they are not registered. (issue #9811 reported by eliel)
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2007-05-28 23:28:52 +00:00
39e9b3112c
Due to the way stringfields work the value of the url pointer will always be non-NULL so we have to use ast_strlen_zero to make sure it is not empty. (issue #9821 reported by pj)
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2007-05-28 23:24:04 +00:00
4b3a3fb14c
Add a new API call for creating detached threads. Then, go replace all of the
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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2007-05-24 18:30:19 +00:00
202fbe363a
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r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines
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r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines
Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)
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2007-05-24 14:43:49 +00:00
9b95428cce
Issue #8409 and accidentally a fix to chan_sip that wasn't supposed to be there
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but is still ok... Sorry. Lack of Tea, really.
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2007-05-24 14:41:43 +00:00
0ec502099f
Yes Virginia, there is a reason why we have stringfields in the sip_pvt structure...
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2007-05-24 11:38:20 +00:00
d1ba4f90de
- Remove debug variable that was only used in one place
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- convert string handling to the ast_str API
- Convert strdup() to ast_strdup() and check the result
- Minor formatting changes
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2007-05-24 03:28:39 +00:00
04e45cfda3
Add SendURL support
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2007-05-24 02:23:08 +00:00
5dc23536ec
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r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines
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r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines
ensure that variables are set on a newly created channel before we start a PBX on it
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2007-05-23 20:53:10 +00:00
a3f9350ec2
Related to issue #9235 btw.
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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines
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r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines
Not getting an ACK to a 200 OK in the initial invite is critical to the call.
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2007-05-18 18:18:59 +00:00
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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines
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r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines
Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.
A special Thank You to WeBRainstorm that gave me access to his system.
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r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 lines
Issue 9487 - stop media flows at hangup of call
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2007-05-18 10:41:31 +00:00
ac343d43c8
Makeup, darling.
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2007-05-18 10:28:56 +00:00
f3ec447a23
Another fix for the support for recordings controlled by INFO-packets
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We still lack a setting to enable/disable this per peer
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2007-05-18 08:49:34 +00:00
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r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines
Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)
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2007-05-17 16:11:26 +00:00
d83dcae6b1
Below patches with some re-structuring for trunk
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r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 lines
Issue #9681 - Handle www-auth on BYE
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2007-05-16 10:58:22 +00:00
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r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines
Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
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2007-05-16 10:09:42 +00:00
c352f7b0d5
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r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines
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r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines
Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!)
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2007-05-16 10:02:06 +00:00
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r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, 16 May 2007) | 17 lines
Merged following patch with a lot of changes for 1.4
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r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines
Issue #9726 - rlister - Better logging for ACL denials
While at it, also added better logging and handling of peers that are not supposed to register.
My patch, stole the issue report from Russell. My apologies, Russell :-)
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2007-05-16 08:51:39 +00:00
7532b0bc4b
Issue #9304 - Update help text to match functionality. Patch by kshumard with changes by oej
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2007-05-16 07:58:43 +00:00
90bad9d2f5
Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable
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2007-05-16 07:35:56 +00:00
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r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 lines
Change -2 to XMIT_ERROR to clarify a bit more
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2007-05-14 19:35:58 +00:00
4fbb449444
If no port is specified in the outboundproxy setting then use the standard SIP port. (issue #9665 reported by tootai)
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2007-05-14 18:21:30 +00:00
fc057b1890
Improve handling network errors on transmission to hosts that don't reply or are unreachable
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With this code, the call will fail as soon as we get a network error. This may happen on
first xmit or a later one, so the retransmit code handles this too.
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2007-05-13 19:20:36 +00:00
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r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2 lines
This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?
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2007-05-12 22:28:09 +00:00
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r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2 lines
Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.
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2007-05-12 21:12:18 +00:00
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r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2 lines
Ensure the onhold flag is set no matter what when being put on hold.
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r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines
Merged revisions 63748 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines
Do not allocate SIP pvt's for PEERs we can not reach.
This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-10 20:51:59 +00:00
7e10164e20
Merged revisions 63611 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines
Merged revisions 63610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines
Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 16:56:45 +00:00
c358b18a5a
Merged revisions 63532 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 lines
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 13:07:44 +00:00
314c874d7d
I noted this on the dev list but got no response, so I just did it myself.
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Lock the call features when being used in chan_sip.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-08 16:41:35 +00:00
d326d84ae0
- Adding some missing spaces
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- Correcting error messages
- Disabling code that doesn't do anything
- Making sure we always respond to this request, happily
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-05 08:05:38 +00:00
02337303ef
a small upgrade to the coding standard, and an update to the code that triggered the upgrade.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 17:49:20 +00:00
3ee0077f04
Added a small bit of code to support the SNOM 360's Record button. Made the find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 16:37:23 +00:00
1b15d8852d
Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that
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is needed to follow the call through the PBX.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 13:56:25 +00:00
81cade7a4c
Merged revisions 62989 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines
Merged revisions 62987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines
When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03 16:45:39 +00:00
e1ec3f917c
Add a small message that we're doing something. On my systems, there's a long
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dead period with a non-responsive CLI after I issue "load chan_sip.so"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 12:12:02 +00:00
1d51b2e161
More username body parts to fix... If working, this needs to be backported to 1.2, 1.4.
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But first, some serious SIP testing :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 12:00:03 +00:00
8fee67c83b
Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properly
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 09:41:03 +00:00
Luigi Rizzo
Joshua Colp
Steve Murphy
Tilghman Lesher
Olle Johansson
Russell Bryant
Kevin P. Fleming
Jason Parker
Mark Spencer