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(issue #7189, Mithraen)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r28896 | kpfleming | 2006-05-19 19:55:31 -0500 (Fri, 19 May 2006) | 2 lines
don't try to predict where the compiler will place things on the stack... put them in the right place explicitly (issues #7029 and #7100, maybe others)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r28794 | kpfleming | 2006-05-19 14:39:55 -0500 (Fri, 19 May 2006) | 2 lines
use the specified 'subscribecontext' for a peer rather than the context found via the target domain (domain contexts are for calls, not for subscriptions) (issue #7122, reported by raarts)
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messagecount2 to messagecount.
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by ppyy)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r28698 | file | 2006-05-19 14:04:02 -0300 (Fri, 19 May 2006) | 2 lines
Make the minidle option actually exist as documented (issue #7159 reported by imran)
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(issue #6866, flefoll)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r28380 | kpfleming | 2006-05-18 15:24:07 -0500 (Thu, 18 May 2006) | 2 lines
handle incoming multipart/mixed message bodies in SIP and find the SDP, if presnet (issue #7124 reported and patched by eborgstrom, but very different fix)
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r28384 | kpfleming | 2006-05-18 15:43:42 -0500 (Thu, 18 May 2006) | 2 lines
fix up a few more places to find the SDP properly (fallout from fix for #7124)
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media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them)
also, documented the 'canreinvite=update' option in the sample config file
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contexts when peer registers (issue #6869 reported by and created by Marquis)
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#7163 reported by and fixed by akohlsmith)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r27973 | file | 2006-05-17 19:34:08 -0300 (Wed, 17 May 2006) | 2 lines
Fix codec priority stuff during authentication (issue #6194 reported by jkoopmann)
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This is a bad fix, since peer should never be 0. Why is it zero
on Royk's system?
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and other countries. (issue #3596 reported by deon patch by dbowerman with minor mods by moi)
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disconnect indications again
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allow native bridging of RTP sessions that are not carrying DTMF even when the bridge needs to listen to DTMF (when SIP INFO is used for DTMF, for example)
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want two streams.
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removed redundant buffer betweend mISDN reading thread and ast_read in favour of the already existing pipe, this clarifies the way a voice frame takes between mISDN and asterisk a lot. centralized debugging of NumberPlan. removed a compiler warning.
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defined in buildopts.h. Also, remove a few more stray spaces in the gcc commands.
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So, I have removed all of the uses of AST_LIST_HEAD_INIT and replaced them
with the equivalent static initializations.
- On passing, fix a memory leak in the unload_module() function of chan_agent.
The agents list mutex was never destroyed, and the elements in the agents
list were not freed.
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for which i need to investigate a bit more (but i am almost 100%
sure that one is also useless).
I do the commit in two steps so it is easier to track the problem.
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- Optionally send systemname in manager (cool when you have a manager proxy)
- Use systemname in CLI prompt
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As the rest of the chan_sip.c changes i am committing, this is
Approved by: oej
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(when the list becomes an astobj this will be revisited;
however at the moment the change would be too intrusive).
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code which called ast_skip_blanks erroneously.
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r26760 | kpfleming | 2006-05-11 03:52:46 -0500 (Thu, 11 May 2006) | 2 lines
ensure that we send a response to REGISTER requests that are successfully authenticated but contain invalid Contact URIs
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(Well, could have opened a branch for this, but felt community pressure not to... ;-)
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