Age | Commit message (Collapse) | Author |
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines
(closes issue #10437)
Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
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something else
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in sip.conf. chan_sip now maintains separate internal MWI subscriptions for
each one.
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the mailbox context. Now, all related MWI event dealings pay attention
to the context as well.
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scalability and is done in such a way that we should be able to add support for other poll() replacements.
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unitialized. Thanks to mvanbaak for pointing this out.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78103 | mmichelson | 2007-08-03 15:25:22 -0500 (Fri, 03 Aug 2007) | 7 lines
Changed the behavior of sip's realtime_peer function to match the corresponding way of matching for non-realtime peers.
Now matches are made on both the IP address and port number, or if the insecure setting is set to "port" then just match on the
IP address.
In order to accomplish this, I also added a new API call, ast_category_root, which returns the first variable of an ast_category struct
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78182 | file | 2007-08-06 13:32:44 -0300 (Mon, 06 Aug 2007) | 2 lines
It is possible for a transfer to occur before the remote device has our tag in which case they send none in the transfer. In this case we need to not fail the transfer dialog lookup.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78139 | tilghman | 2007-08-04 22:29:01 -0500 (Sat, 04 Aug 2007) | 2 lines
If peer is not found, the error message is misleading (should be peer not found, not ACL failure)
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also need to be macros if DETECT_DEADLOCKS is set
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77824 | mmichelson | 2007-07-31 10:21:22 -0500 (Tue, 31 Jul 2007) | 6 lines
This patch makes Asterisk send 100 Trying provisional responses upon receipt of re-invites. This makes it so that if there are two or more Asterisk
servers between endpoints, the Asterisk servers will not keep retransmitting the re-invites.
(closes issue #10274, reported by cstadlmann, patched by me with approval from file)
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locking info when DEBUG_THREADS is enabled... i could have fixed this to be dependent on DEBUG_THREADS, but it would be just as easy for someone to add their test/debugging code to the macros as it would have been to the functions
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old one fails to compile on my 64-bit system with gcc-4.1 and --enable-dev-mode turned on
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calls with sip_pvt_lock and sip_pvt_unlock, you lose the tracing info you would normally get via DETECT_DEADLOCKS; so I turn these two functions into macros when DETECT_DEADLOCKS is called. This way, you get meaningful stuff in the file and func slots in the lock_info struct.
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+ place the link field at the beginning of struct sip_pvt,
and not somewhere in the middle;
+ in __sip_reliable_xmit, remove a duplicate assignment, and
put the statements in a more logical order (i.e. first copy
the payload and associated info, then copy arguments from the
caller, then finish initializing the headers...)
nothing to backport.
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was misleading - the function deals with all incoming packets, be
them requests or responses.
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SIP_NOVIDEO, SIP_DIALOG_ANSWEREDELSEWHERE, SIP_PAGE2_NOTEXT,
SIP_PAGE2_OUTGOING_CALL
These are seldom used so the diff is relatively small.
Note that 'OUTGOING_CALL' is dangerously similar to another
dialog flag, 'SIP_OUTGOING', so the description will need to
clarify the different meaning of the two.
Also note that the description of NOTEXT is a bit unclear - does
it mean we don't support it, or 'not requested or not supported' ?
On passing fix a comment referring to video instead of text.
Finally, mark with XXX a possibly misleading debugging message.
(maybe the latter is worth backporting).
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CLI lines. This helps maintaining consistency on output, slightly
improves readability, and maybe one day will make it easier to
translate the output in other languages (though i have a hard time
believing that a CLI user who needs 'yes' and 'no' to be translated
can actually figure out what he/she is doing!)
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Start putting these variables in a single struct (called 'sip_cfg' for the time
being, but it could as well be 'global' or some other name) so it
is easy, when reading the code, to figure out what they are for.
The downside of using struct fields instead of individual global
variables is that the compiler cannot tell if there are unused fields.
But the advantage of not cluttering the namespace and manilpulating
all these variables at once certainly overcome the disadvantagess.
Nothing to backport, again.
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at load time instead of duplicating the initializer.
This should remove the risk of forgetting fields in the
initializer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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use AST_FORMAT_AUDIO_MASK instead of playing with AST_FORMAT_MAX_AUDIO
to determine audio formats.
There is a dubious use of AST_FORMAT_MAX_AUDIO in sip_request_call()
which surely needs fixing, namely:
/* mask request with some set of allowed formats.
* XXX this needs to be fixed.
* The original code uses AST_FORMAT_AUDIO_MASK, but it is
* unclear what to use here. We have global_capabilities, which is
* configured from sip.conf, and sip_tech.capabilities, which is
* hardwired to all audio formats.
*/
The latter is possibly something to backport when fixed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Move together flags used in the same way (e.g. dialog only,
dialog-peer, ...) so it will become easier to deal with them
in a more systematic way.
This is being done in stages so it will be easier to detect
breakage, if any should occur.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Remove definitions for now-unused flags, and add references to print
routines for other flags.
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the original pointer while calling the function.
on passing add some comments on one of the places where it
is used, and explain why it is safe there.
again, a no-op for practical purposes.
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dialog_ref/unref (they are a no-op at the moment).
Also clean a pointer after freeing memory to avoid
dangling references, and write a for() loop in canonical form.
In practice, everything in this commit is a no-op.
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This commit is, for all practical purposes, a no-op, as it only
introduces the dialog_ref() and dialog_unref() methods, and uses them
in a few places (not all the places where they would be needed).
The goal is to start annotating the code with these calls, so the transition
to a proper container will be easier.
Nothing to backport.
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In a nutshell, these fields are used to tell a sip entity
the address and port its request came from, and are extremely
useful in the presence of NATs, especially with symmetric NATs
where STUN is totally ineffective.
This patch stores the address and port in the 'ourip' field of
the dialog descriptor, so they can be reused in subsequent transactions.
As it is, it works well for things like REGISTER requiring authentication,
because the second REGISTER request (with auth credentials) will carry
the correct address. Maybe it can also be useful, in case of an address
change, to do one or both of the following:
+ propagate the new address to the parent user/peer descriptor so that new
dialogs will use the correct address from the beginning.
This is trivial to implement, I am just waiting for feedback on this.
+ re-issue a request in case of an address change. This a lot less trivial,
maybe unnecessary, and probably covered by the previous item.
I would seriously consider this patch for addition to 1.4 and 1.2.
The code is very little intrusive, and it would solve in a correct
way the nat traversal problems for which externip/externaddr/stunaddr
are only a partial and expensive workaround.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77536 | file | 2007-07-27 13:27:16 -0300 (Fri, 27 Jul 2007) | 6 lines
(closes issue #10323)
Reported by: julianjm
Patches:
chan_sip_device_state_hold_fix.v1.diff.txt uploaded by julianjm (license 99)
Clear ONHOLD flag when decrementing the onHold peer count. If we did not do this the count may keep decreasing.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77490 | mmichelson | 2007-07-27 09:30:43 -0500 (Fri, 27 Jul 2007) | 3 lines
"re-invite" was misspelled
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(closes issue #10277, patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
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* random() is defined to return a positive result, and now ast_random()
will always do so as well
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(closes #10308)
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+ externexpire was not initialized properly;
+ stunaddr was not handled properly on a sip reload
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does not use DTMF BEGIN frames.
1.4 seems correct (it does not have the two fields).
However, as this bug shows, the current way of creating the sip_tech
replica is too error-prone, one can easily forget to update one of
the two entries. Perhaps it would be better to create sip_tech_info
expliclty at module load, by doing
sip_tech_info = sip_tech;
sip_tech_info.send_digit_begin = NULL
(in this case, this is something applicable to 1.4 as well).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76561 | file | 2007-07-23 11:34:21 -0300 (Mon, 23 Jul 2007) | 14 lines
Merged revisions 76560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 lines
(closes issue #10236)
Reported by: homesick
Patches:
rpid_1.4_75840.patch uploaded by homesick (license 91)
Accept Remote Party ID on guest calls.
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allow building under dev mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by: snuffy
Patches:
doxygen-updates.diff uploaded by snuffy (license 35)
Another big batch of doxygen documentation updates
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between integers and strings using a single translation table,
and use them in a few places instead of ad-hoc routines
that duplicate the table.
On passing, note that REFER_CONFIRMED is never used, and add a
few comments.
Nothing to backport here.
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handling of the sip_register structures.
This commit only changes comments and whitespace.
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