Age | Commit message (Collapse) | Author |
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use an enum for authentication results and clean up code
fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r29764 | kpfleming | 2006-05-23 13:16:40 -0500 (Tue, 23 May 2006) | 2 lines
simplify/fix lock retry, and fix comment
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lock that may have been obtained elsewhere. Prevents the monitor thread of the SIP module from going into an infinite loop, effectively, breaking SIP until you restart Asterisk or the mutex is unlocked, whichever comes first.
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with dubious assignments.
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usage and to ease understanding of the function themselves.
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Note that in a couple of places the code overwrites the string
in the buffer - mark the two places with XXX but do not fix the
bug yet.
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These are just the first two instances, there are many more to replace.
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initialization, an extra variable (newpeername) and a
variable newpeername, and an 'else' branch.
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On passing simplify an ast_cli message nearby.
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(issue #7189, Mithraen)
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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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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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This is a bad fix, since peer should never be 0. Why is it zero
on Royk's system?
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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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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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