asterisk

Commit Graph

Author	SHA1	Message	Date
Jason Parker	788e66cf20	Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	d6eabdc0dd	fix this line again git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	fb475c8c7e	clean this up, sorry my brain is not really working git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306964 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	8bd2bfb587	Backup file storing message duration is not used with IMAP_STORAGE, remove code. The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	e9dcd4a9e2	make this safer and fully correct, pointed out by Steve Davis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	9fe4bc60b5	Minor AST_FRAME_TEXT related issues. * Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jason Parker	e03c6ed738	Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305252 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3c5fd20e90	Fix compilation when ODBC_STORAGE is defined. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	1fb86f03c8	Backport MeetMe related reference leaks fixes from 1.6.2/1.8/trunk. I had forgotten that MeetMe in 1.4 also used astobj2, so backport the fixes where appropriate. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b30fe5496f	Fix voicemail sequencing for file based storage. A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303676 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	58a666ac5a	Fix channel redirect out of MeetMe() and other issues with channel softhangup. Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	2c54f15171	Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	9f2050cc3d	Option L() is milliseconds, not seconds. Change the verbose output of option L() to say milliseconds and not seconds as the value is in milliseconds. (closes issue #18264) Reported by: jacco Patches: app_dial_patch.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jacco git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@302916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	25181e5bdb	change variable name to the same used in later branches git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@300924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	93864bd03d	Ensure good bye prompt in voicemail is played at the correct time. Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@300918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	24118f5839	Typos: recieved => received git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@299002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	ccfbb0a30a	After recording only silence for a voicemail prepending, restore backup files. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	71115232da	Fix improper hangup when doing an attended transfer to queue. Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298596 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	4369b3c4ae	Don't create a Local channel if the target extension does not exist. (closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297689 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	92ea2179e5	Add "DAHDI" to a couple of app_meetme error messages. This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	25c2ccaac3	Properly restore backup information file when hanging up during message prepending. ABE-2654 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	72d3cd8b47	Handle failures building translation paths more effectively. The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	d32582d42f	The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	3fc98d5530	Ensure original message duration is preserved when prepending a message. It seems the fix to issue 17103 was a little overzealous and removed the code that backed up the textfile containing the original message duration. This code has now been restored. (related to issue #17103) ABE-2654 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	fe1ca1a8da	Fix regression causing abort in voicemail after opening a mailbox with no mesgs. In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	969aac85dd	Fix inprocess_container in voicemail to correctly restrict max messages. The comparison function logic was off, so the number of sessions for a given mailbox were not being incremented properly. This problem caused the maximum number of messages per folder to not be respected when simultaneously leaving multiple voicemails just below the threshold. These problems should be fixed by the above, but just in case: Fixed resequence_mailbox to rely on the actual number of detected number of files in a directory rather than just assuming only 10 messages more than the maximum had been left. Also if more messages than the maximum are deleted they are actually removed now. The second purpose of this commit should have been separated out probably, but is related to the above. Again, if the number of messages in a given voicemail folder exceeds the maximum set limit make sure to allocate enough space for the deleted and heard index tracking array. A few random fixes: There was a forgotten decrement of the inprocess count in imap_store_file. When using IMAP storage, do not look in the directory where file based storage messages may still reside and influence the message count. Ensure to use only the first format in sendmail. ABE-2516 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293004 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	c560e9994b	Record priv-recordintro as sln, not gsm This removes the gsm->sln step when transcoding priv-recordintro. (closes issue #18176) Reported by: pabelanger Patches: chan_sip.diff uploaded by pabelanger (license 224) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@292411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	c7fd3b73d3	Fix improper operator key acceptance and clean up temp recording files. This is a fix for when pressing the operator key after recording an unavailable, busy, name, or temporary message in mailbox options. The operator key should not be accepted here, but should be allowed during the message recording. If the operator key is pressed during ensure the file is saved or deleted as apporopriate. Also, ensure removal of temporary recorded files after an early hang up or when message acceptance confirmation times out. ABE-2518 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@292223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	e93fc235b1	When forwarding a message, a prepend means that the filesystem will always have a better copy. (closes issue #17803) Reported by: dpetersen Patches: 20100923__issue17803.diff.txt uploaded by tilghman (license 14) Tested by: dpetersen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	c4f3f68e43	Fix a crash in app_sms. Since the data being passed to the generator callback is on the stack of the SMS() application, we must ensure that the generator is stopped before the application exits. ABE-2587 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	3702c0cc02	Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a user and admin pin setup for your conference, using the user pin would gain you admin priviledges. Also, when no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the user tried to enter a conference then they were still prompted for a pin and forced to hit #. (closes issue #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: kuj Review: [full review board URL with trailing slash] git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@287758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	41969111bd	Blank columns should get set on reload, not ignored. (closes issue #16893) Reported by: haakon Patches: 20100818__issue16893.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@287386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	a91c96ba5d	whitespace fix git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	fa2b4c0196	Ensure mailbox is not filled to capacity before doing message forwarding. Specifically, before prompting to record a prepended message the capacity is checked first. If the mailbox is full the extension will be reprompted. ABE-2517 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@286941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Brett Bryant	1811246e48	Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages. (closes issue #15726) Reported by: 298 Patches: M15726.diff uploaded by junky (license 177) Tested by: junky Review: [full review board URL with trailing slash] git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@285194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	a2bf7447e2	Properly detect when a sound file doesn't exist ast_fileexists returns -1 for error and 0 for a non-existant file. The existing code treated missing files as though they were existed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@284881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	1c72a8ab49	Reset visible indication after answer. (closes issue #17641) Reported by: klaus3000 Patches: ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65) Tested by: schmidts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@281566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jean Galarneau	cf0d36226d	Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@280341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	08883af231	SIP promiscuous redirect could fail to dial the redirect. The ast_channel was created with one variable to ast_request() but the call to ast_call() that initiates the outgoing call was using a different variable. The two variables are not equivalent if the call_forward string included a channel technology specifier. e.g., SIP/200 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@279206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7df9bcf5e7	Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message. (closes issue #16350) Reported by: noahisaac Patches: 20100623__issue16350.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@278261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	d9e34da4ae	Total analysis time error with SIP and silence suppression When using app_amd with SIP providers that have silence suppression on, the iTotalTime count increases exponentially. (closes issue #17656) Reported by: juls git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@277182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	c66d95c8e5	Make user removals and traversals thread safe in meetme. Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	fa79e11a91	Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial (closes issue #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@275027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	989ecebcb0	Make get_member_status return QUEUE_NO_MEMBERS instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work again. This regression was introduced in 273639. Also fixed whitespace. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@274093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	97f61244f9	Fix various typos, reported by Lintian git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273640 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	358159c2b0	If all members are paused, the wrong status is indicated. (closes issue #17576) Reported by: ramonpeek Patches: diff.txt uploaded by ramonpeek (license 266) Tested by: ramonpeek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	9bae102b7c	Allow admin user to join conference without using admin mode and no user pin. Configuring the conference in meetme.conf like the following: conf => 2345,,6666 did not prompt for pin when used without admin mode. This meant that the conference could not be joined as an admin even if the user knew the correct pin. The original bug report was submitted claiming that the blank user pin should deny entry into the conference. I think a better way to handle this would be with a feature enhancement that used the following syntax: conf => 2345,X,6666 - where X denotes no acceptable pin allowed (closes issue #15704) Reported by: modelnine git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273474 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	f5310e3bc7	Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. Specifically checking for this scenario by looking at the appropriate control frames resolves the issue. (closes issue #15871) Reported by: Ivan Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@273354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	d1292255d1	Send AgentComplete manager events in the event of blind and attended transfers. (closes issue #16819) Reported by: elbriga Patches: app_queue.diff uploaded by elbriga (license 482) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	2f6313a1c8	First caller into a dynamic conference now enter pin once. If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@272255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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2836 Commits (e90101cecf75a650afb4d51b0b9bd0085cc55418)