The name "ser" was used in a lot of places. However, it is a relic from when
the struct was a server_instance, not a session_instance. It was renamed since
it represents both a server or client connection.
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r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines
Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI.
(closes issue #12560)
Reported by: vsauer
Patches:
patch001.diff uploaded by ramonpeek (license 266)
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r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines
When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0.
(closes issue #13599)
Reported by: hjourdain
Patches:
chan_sip.c.diff uploaded by hjourdain (license 583)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162739 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Terry Wilson created the original patch for this functionality, which I slightly modified and added
the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not
do anything for G711 over SIP fax detection. By default, this option is disabled.
Reviewboard: http://reviewboard.digium.com/r/69/
This functionality is for issue AST-140.
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r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines
Jon Bonilla (Manwe) pointed out on the -dev list:
"I guess that having only ip-phones in mind is not a good approach. Since it is
possible to have a sip proxy connected to asterisk we could receive a 407
(unauthorized) or 483 (too many hops) as response and dialog ending would not be
a good behavior."
So modified.
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r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines
When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion
fails, and the resulting integer is garbage. Thus, we must initialize the
integer and check it afterwards for success.
(closes issue #14000)
Reported by: folke
Patches:
asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626)
asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626)
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------------------------------------------------------------------------
r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines
update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
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in addition:
move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
it would be best to maintain API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to ao2_callback() except
that it allows you to pass arbitrary data to the callback.
Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
is only found if it dialed the extension that was subscribed to. You can now
specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as
it's value implies, ignore the current context of the caller when doing the
lookup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
on 32-bit systems, so we need to use %llu instead. Of course
%llu is 128-bits on 64-bit systems, so we have to cast to
unsigned long long. No harm, but it's sure annoying.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
from an int to a uint64_t. This prevents potential comparison
problems from happening if the version string exceeds
INT_MAX. This was an apparent problem for one user who could
not properly place a call on hold since the version in the
SDP of the re-INVITE to place the call on hold greatly
exceeded INT_MAX.
This also aligns with RFC 2327 better since it recommends
using an NTP timestamp for the version (which is a
64-bit number).
(closes issue #13531)
Reported by: sgofferj
Patches:
13531.patch uploaded by putnopvut (license 60)
Tested by: sgofferj
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r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines
We don't handle 4XX responses to BYE well. According to
section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
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r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines
Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
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r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines
Add some missing invite state changes necessary in the sip_write
function. Not setting the invite state correctly on the call was
resulting in the Record application leaving empty files. I also
have updated the doxygen comment next to the declaration of the
INV_EARLY_MEDIA constant to reflect that we also use this state
when we *send* a 18X response to an INVITE.
(closes issue #13878)
Reported by: nahuelgreco
Patches:
sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162)
Tested by: putnopvut
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
changed from using a global lock in update_call_counter to
using the locks within the sip_pvt and sip_peer structures
instead.
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* Revert logic to mirror 1.4's in the sense that it will not allow
the call counter to dip below 0.
These two measures prevent potential races that could cause a SIP peer
to appear to be busy forever.
(closes issue #13668)
Reported by: mjc
Patches:
hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
removed a very important line to set the "len" field
for incoming SIP requests. The result was that all incoming
SIP messages appeared to be 0-length, meaning Asterisk
could do no meaningful processing of anything SIP-related
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- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
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ast_channel_search_locked need to be made. Specifically, the caller needs to be able to
pass arbitrary data which in turn is passed to the callback. This patch addresses all
of the nested functions currently in asterisk trunk.
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makes more sense. Where it was set before, it was impossible
to actually delay sending a CANCEL if we had not yet received
a provisional response to an INVITE.
(closes issue #13626)
Reported by: atis
Patches:
13626.patch uploaded by putnopvut (license 60)
Tested by: atis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either
of these mandates that the passed 'arg' is a hashable object, making
searching for an ao2 object based on outside criteria difficult.
Reviewed by Russell and Mark M. via ReviewBoard:
http://reviewboard.digium.com/r/36/
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container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would
only remove a single object.
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channel list calling a caller-defined callback. The callback returns non-zero
if a match is found. This should speed up some of the code that I committed
earlier today in chan_sip (which is also updated by this commit).
Reviewed by russellb and kpfleming via ReviewBoard:
http://reviewboard.digium.com/r/28/
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the calling party when subscribed to the state of an extension that is ringing.
This has some limitations which are documented in sip.conf.sample.
(closes issue #13827)
Reported by: seanbright
Patches:
issue13827.patch uploaded by seanbright (license 71)
Reviewed by: russellb
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remotesecret => our password for a remote service
secret => our authentication when someone calls us
Secret => still has both functions if remotesecret is not used.
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