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r30033 | file | 2006-05-24 16:14:01 -0300 (Wed, 24 May 2006) | 2 lines

Fix deadlock caused by a race condition in the logger when reloading (issue #7195 reported and fixed by softins)

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r29973 | kpfleming | 2006-05-24 11:59:20 -0500 (Wed, 24 May 2006) | 2 lines

support video recording via AGI 'RECORD FILE' command (issue #7068)

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r29971 | kpfleming | 2006-05-24 11:52:08 -0500 (Wed, 24 May 2006) | 2 lines

fix various bugs related to exiting from queue via keypress and moh handling (issue #6776, different fix)

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r29969 | kpfleming | 2006-05-24 11:17:26 -0500 (Wed, 24 May 2006) | 2 lines

respect 'usecallingpres' in zapata.conf even if CLID has not been set for the channel (issue #7123)

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fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars

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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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(issue #7190, Mithraen)


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r29849 | russell | 2006-05-23 17:44:52 -0400 (Tue, 23 May 2006) | 2 lines

fix the sourceaddress option (issue #7213, alphaque)

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later on.


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patch 0006763 with minor changes.


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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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 Sanity check code for an extended failure in trying to obtain a channel 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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update ast_mutex_init to allow mutexes that are all zero bytes to be initialized (in the case of a dynamically-allocated structure containing a mutex)


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gets a chance to.


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r29696 | bweschke | 2006-05-23 11:58:24 -0400 (Tue, 23 May 2006) | 3 lines

 Fix a potential leak and correct (hopefully) a segfault under certain conditions. #6784 (vovan and perry testing)


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  app_meetme Muting and Manager API enhancements #6731 (softins w/some minor mods to accomodate recent enum work)



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* when receiving a connect from the NT Side we wait until we have the final
  l3id until we queue the answer to asterisk to avoid bridging conflicts
* when not bridged to misdn we had a segfault after receiving the connect 
  due to a strcasecmp bug.. this didn't happen before, cause we hadn't had
  the bridge before
* cleanup of the bchannels is queued now, due to possible race conditions
* added mISDN_clear_stack when cleaning the bchannel



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manager naming convention.


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r29555 | file | 2006-05-22 18:27:12 -0300 (Mon, 22 May 2006) | 2 lines

Increase the silence threshold to 128 to "fix" it, so I'm told. (issue #6595 reported by davetroy fixed by casper)

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r29512 | file | 2006-05-22 17:15:04 -0300 (Mon, 22 May 2006) | 2 lines

Use the correct language when playing the transfer sound (issue #7109 reported by casper)

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