versions of the format string are identical. Also, since each format is only
used once, get rid of the use of defines all together. (issue #8344, julieng)
In passing, also clean up the formatting a but to get rid of the nesting
without the use of braces, as defined in the coding guidelines.
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- Keep RTP running during T.38 session
We might improve the code to issue ast_rtp_stop if T.38 re-invite not fails
later on in the code, but I don't see many reasons to.
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r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) | 7 lines
Work around an issue that caused menuselect to display a bogus description for
app_voicemail and chan_zap. These modules use some preprocessor directives to
determine what it will report to Asterisk as its description. However, the way
we extract this information from the source files for menuselect is not smart
enough to figure this out.
(issue #8326, #8328)
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r47380 | file | 2006-11-09 11:53:25 -0500 (Thu, 09 Nov 2006) | 10 lines
Merged revisions 47379 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 lines
Don't include compiler.h on kernels 2.6.18 and higher as, well, it's apparently going to be removed. This should make all you FC6 fans happy as your Asterisk will now build without any mods.
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- Fix documentation for sip_pvt_lock/unlock - doxygen does not inherit like zapata.conf !!!
- Change doc for a sip_pvt setting
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avoid doing p > 0 when p is a pointer;
move a lock closer to the place where it is needed
Approved By: oej
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Same as for peers and users, replace ASTOBJ_UNREF(r, sip_registry_destroy)
with unref_registry(r);
Approved By: oej
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Replace ASTOBJ_UNREF(peer, sip_destroy_peer) with unref_peer(peer);
This places the name of the destructor in one place only (where it
should be), eliminates the chance of errors in case you specify the wrong
destructor, and also lets the compiler do type checking on the argument,
again helping with keeping the code clean.
Same for users.
remove two duplicate definitions.
Approved By: oej
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Overall i think the previous change to ast_channel_alloc()
to close bug 7506 should have been done by defining
an ast_set_callerid_noevent() function that does the
setting without generating the event.
Lot less code duplication, and easier to handle.
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to the peer side of a type=friend.
This is for trying to support BJ in his quest to solve some issues
with the queue system and type=friend objects.
BJ: Please test!
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I think this module doesn't compile, anyways, because
it has not been updated to the new module interface.
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e.g. in the Manager interface. This information is available as
a callerid (or something like that) during a call, but not when a
device is registered but silent.
It may be useful to have it available e.g. when developing a user
interface/operator panel, to map numbers to names.
experimental, so not committed to 1.4
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r46775 | file | 2006-11-01 13:21:34 -0500 (Wed, 01 Nov 2006) | 2 lines
It's another round of chan_iax2 fixes! Should hopefully fix the deadlock issues people have been reporting. IAXtel now has qualify turned on for 800 peers and it is handling it fine.
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is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
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- removing transmit_reinvite_with_t38_sdp in favour of adding an argument to
transmit_reinvite_with_sdp
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lock when needed - when we remove the dialog from the dialog list
If this doesn't lead to severe problems, it might help with some locking issues
in 1.4/1.2.
- Remove the term "interface" as a synonym for a SIP dialog. Sorry, Mark, but no
one understands it... ;-)
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r46358 | russell | 2006-10-27 10:32:40 -0500 (Fri, 27 Oct 2006) | 5 lines
Instead of iterating all of the options once to look for jitterbuffer options,
and then again for everything else, move the processing of jitterbuffer
options into the main loop so that there are no erroneous messages about
ignoring unknown options. (issue #8226)
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r46351 | crichter | 2006-10-27 11:49:20 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line
added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session
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r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line
fixed not compile issue, which was just introduced
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r46353 | crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line
fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c
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r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006) | 2 lines
add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using
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r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006) | 2 lines
ensure that the translation matrix is properly lock-protected every place it is used
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r46152 | kpfleming | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines
if multiple translators are registered for the same source/dest combination, ensure that the lowest-cost one is always inserted earlier in the list
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r46153 | kpfleming | 2006-10-24 19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines
code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable
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when the number of channels fill the MTU on a given link.
In the future, this needs to be configurable per peer with trunking enabled.
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appreciated really. (Read the coding guidelines).
I've worked hard to make chan_sip a better place to code in, let's
keep it that way and don't add more stuff without comments.
Thank you.
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so fix the places where this might happen.
This is also a fix that ought to go into 1.4
[The difference between the two functions is a bit confusing,
and in asterisk i believe all string handling functions
should be able to handl a NULL string as argument,
but changing the API in trunk and not in 1.4 would make
backporting harder.]
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As discussed on the mailing lists, 0 is a legal value
for Cseq, so there is no point to treat it specially.
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for NULL is certainly wrong and usually disables the
checks that we want to make instead.
This commit fixes a number of the above bugs where the result
of get_header() is immediately checked for NULL.
This is certainly a candidate for merging into 1.4
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On passing, remove a wrong comment (that probably I wrote
myself!) and introduce a temporary variable to avoid a
misleading cast.
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simplifies its usage.
+ add another client for parse_uri, in handling Contact: strings
(on passing, document the content of the "fullcontact" field);
+ in register_verify(), mark with XXX what i believe is another
misinterpretation on the URI format when '@' is missing.
No code changed here, so no fixes applied.
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fix parse_uri() to interpret a missing userinfo
section as a domain-only URI, and comment a wrong
interpretation of the above in check_user_full().
The function has been patched to preserve the existing
behaviour (in what admittedly is a corner case, but
could be received under attacks).
Hopefully the From: based matching will go away soon!
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before splitting around the @, otherwise the refer_to_domain
might contain arguments as well, causing failures.
I think this is a true bug that ought to be fixed in 1.4 as well.
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introducing the function parse_uri() that splits
a URI in its components.
Right now use it only in one place, because the custom
parsing that is done here and there sometimes has
bugs that i want to figure out first.
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apart from a small (but disabled by default) new option.
In detail:
+ introduce a new value for enum check_auth_result, AUTH_DONT_KNOW,
used (read below) when a function does not have a conclusive response.
Possibly this is the same as AUTH_NOT_FOUND, but need to check further.
+ move the large blocks (checking in the users list and in the peers
list, respectively) from check_user_full() to separate functions.
They return AUTH_DONT_KNOW in case they don't find a match, so
the caller know that it has to try the next method.
There is still some duplication of code here, but i
have not tried yet to remove it.
+ [new option] a new option in sip.conf, match_auth_username,
has been introduced, and disabled by default.
If set, and the incoming request carries authentication info,
the username to match in the users list is taken from there
rather than from the From: field.
This change is easy to identify, being made of
- one line to declare the variable match_auth_username
- a block of 15 lines in check_user_full()
- one line in sip list settings
- two lines for parsing the config file.
check_user_full() is now a lot cleaner - basically a sequence of
checks that are applied to the request. This will help future
work with new matching schemes.
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A large block needs reindentation now, but we don't do that because
it can be moved to a separate function.
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lock these data structures.
This improve readability, and also hides the underlying
locking mechanism so it is a lot easier to add diagnostic
code, or move the object locks somewhere else, etc.
On passing, rename the lock field in sip_pvt to pvt_lock,
also for ease of readability.
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allow custom threadstorage init functions to return failure
use a custom init function for chan_sip's temp_pvt, to improve performance a bit
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be called for each thread specific object after they are allocated. Note that
there was already the ability to define a custom cleanup function. Also, if
the custom cleanup function is used, it *MUST* call free on the thread
specific object at the end. There is no way to have this magically done that
I can think of because the cleanup function registered with the pthread
implementation will only call the function back with a pointer to the
thread specific object, not the parent ast_threadstorage object.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45408 | kpfleming | 2006-10-17 17:24:10 -0500 (Tue, 17 Oct 2006) | 3 lines
optimize the 'quick response' code a bit more... no more malloc() or memset() for each response
expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed
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r45378 | file | 2006-10-17 16:30:34 -0400 (Tue, 17 Oct 2006) | 2 lines
Don't create a "real" pvt structure for requests that shouldn't be able to create one. Instead use a temporary pvt and fill it with enough information so we can send a reply.
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I *know* it is not required, but it makes navigation easier and will help
when splitting up this large source code file.
Thank you!
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necessary - rather, cast the argument to int.
In this case, the string is in a UDP packet and as such
limited to 64k so its length can be safely represented in an int
without truncation (besides, this is just a debugging message!)
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To avoid the same mistake in the future (due to slightly
confusing variable names), add a comment.
On passing, remove a redundant initialization.
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and friends in a function, auth_headers(), which is used to
simplify the interface of do_{proxy|register}_auth().
+ use PROXY_AUTH = 407, WWW_AUTH = 401 as values for enum sip_auth_type;
No functional change, only code cleanup.
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authentication issues. This was committed in revision 44844, where the commit
message was just "small formatting cleanup", so I am pretty sure he didn't mean
to commit this part.
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r44561 | crichter | 2006-10-06 14:50:25 +0200 (Fr, 06 Okt 2006) | 9 lines
Merged revisions 44334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04 Okt 2006) | 1 line
added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible
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and rename the old DEFAULT_SIP_PORT as STANDARD_SIP_PORT
to make it clear that this is not something we can change,
unlike other defaults.
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that, but the protocol clearly states that if we DO NOT mention a port it
is 5060. DEFAULT_SIP_PORT is whatever we default to listen to.
I believe it's the third time I revert a patch like this.
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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().
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r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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r44078 | pcadach | 2006-09-30 22:12:23 +0600 (Сбт, 30 Сен 2006) | 6 lines
Fix issue #7928 correctly. Next is a comment of previous fix:
Issue #7928 - Don't send both 404 and 503. Fix by phsultan with
a small fix by me, myself or I. Thanks, Philippe!
(This was caused by my changes to the transaction handling)
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r44068 | pcadach | 2006-09-30 10:37:39 +0600 (Сбт, 30 Сен 2006) | 14 lines
Found some buggy SIP clients (phones Planet VIP-153T firmware
1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK
message only (when remote party answers) but on RINGING message
too, so when we send 200 OK message, we get unidentified ACK
message (because INVITE acknowledged on RINGING message already),
so 200 OK retransmits within its retransmission interval then
call gets dropped.
If someone else knows how to provide workaround for such cases,
please, fix it in correct way.
Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.
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r44034 | pcadach | 2006-09-30 02:43:13 +0600 (Сбт, 30 Сен 2006) | 1 line
Fake display name by called number on incoming calls (until passing connected number/connected name is not implemented)
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r44042 | pcadach | 2006-09-30 03:05:43 +0600 (Сбт, 30 Сен 2006) | 1 line
Set TON/PRESENTATION information more carefully when no CallingNumber IE available
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r44043 | pcadach | 2006-09-30 03:09:10 +0600 (Сбт, 30 Сен 2006) | 1 line
Compile first, please
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r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) | 1 line
Put attribute tag at correct place
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r43862 | pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line
Force remote side to start media on outgoing PROGRESS message
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r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) | 1 line
Fix ASN1 description of non-standard Cisco extensions
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r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't treat unknown control frames as voice
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r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't warn on HOLD/UNHOLD control frames
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r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28 Сен 2006) | 1 line
Do not open transmit channel until TCS is received
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r43783 | file | 2006-09-27 13:00:31 -0400 (Wed, 27 Sep 2006) | 2 lines
Get rid of two functions from a time now past (we THINK these are from pre-recursive lock time) that may be contributing to two open issues on the bug tracker (7562/7939) and that has the potential to just make bad things happen if the timing is right.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43650 | qwell | 2006-09-26 08:33:47 -0700 (Tue, 26 Sep 2006) | 11 lines
Add proper codec support to chan_skinny. Works with at least ulaw, alaw, and g729a.
This is technically a "new feature", but there are justifications for it.
I found a bug with the recent rtp packetization changes, which caused the media setup to
fail under certain circumstances, particularly when using allow=all, or having no allow=
statements (globally or on the device).
I could have either removed the rtp packetization features, or I could add proper codec
support (which, without, I think most people would consider to be a bug anyways).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43564 | russell | 2006-09-24 10:58:10 -0400 (Sun, 24 Sep 2006) | 5 lines
Fix a CLI command registration issue where an erroneous message claiming that
"iax2 show provisioning" was already registered. This was because this command
was registering itself as both the command, as well as the command it is
deprecating. (issue #8022, reported by bjweeks, fixed by myself)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43553 | russell | 2006-09-24 09:53:35 -0400 (Sun, 24 Sep 2006) | 12 lines
Merged revisions 43552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43552 | russell | 2006-09-24 09:50:30 -0400 (Sun, 24 Sep 2006) | 4 lines
Check to see if the channel that is activating the IAXPEER function is actually
an IAX2 channel before proceeding to process it to avoid crashing.
(issue #8017, reported by admott, fixed by myself)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42133 | bweschke | 2006-09-06 14:16:41 -0400 (Wed, 06 Sep 2006) | 6 lines
Look ma! No more deadlocks! <sic>
As posted from #7458 and others similar to it in Mantis:
p->app_lock was a mutex really designed for use with agents not in callback mode. That being the case, I've tried to code it so that when callback mode is used, the app_lock mutex will not be locked/unlocked at all. Please let me know how you make out - and if you continue to deadlock now, please reproduce the deadlock logging information and post to Mantis.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Russell Bryant
Olle Johansson
Matthew Fredrickson
Joshua Colp
Tilghman Lesher
Steve Murphy
Kevin P. Fleming
Luigi Rizzo
Nadi Sarrar
Matt O'Gorman
Christian Richter
Jason Parker
Paul Cadach
Mark Spencer
BJ Weschke