Review Board 1.7.16


VoicemailMain and VMauthenticate: Like VoiceMail allow escape to the 'a' extension when a single '*' is entered in Mailbox or Password

Review Request #489 - Created Feb. 4, 2010 and submitted

Alec Davis
trunk
Reviewers
asterisk-dev
Asterisk
VoiceMail() already has the ability to escape to the single digit 'a' extension.

Where a site uses VoiceMailMain(mailbox), the users have to be at their own extension to clear their voicemail, they have no way of escaping VoiceMailMain to allow entering there own mailbox.
There are other reasons to require escape also.

The only option currently is to dial another number where VoiceMailMain() is called without the mailbox parmeter, and asks for Mailbox and Password.

This patch, allows a site to include to 'a' extension in the current dialplan context, to allow an escape.

If the 'a' priority doesn't exist in the context VoicemailMain is called from, then it acts as the old behaviour.
Tested both VoiceMailMain and VMauthenticate with the 'a' extension present and not. All 4 senarios performed as expected.
Using the following dialplan:

[voicemail-main]
exten => s,1,Answer()

exten => s,n,VMAuthenticate()
exten => s,n,VoiceMailMain()
exten => s,n,Verbose(0,After VoiceMailMain)
exten => s,n,Hangup

exten => a,1,Verbose(0,User entered *)
exten => a,n,Playback(connecting)
exten => a,n,Goto(s,1)

Diff revision 4 (Latest)

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1 2 3 4

  1. /trunk/UPGRADE.txt: Loading...
  2. /trunk/apps/app_voicemail.c: Loading...
/trunk/UPGRADE.txt
Revision 262004 New Change
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===========================================================
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===========================================================
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===
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===
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=== Information for upgrading between Asterisk 1.6 versions
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=== Information for upgrading between Asterisk 1.6 versions
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===
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===
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=== These files document all the changes that MUST be taken
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=== These files document all the changes that MUST be taken
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=== into account when upgrading between the Asterisk
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=== into account when upgrading between the Asterisk
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=== versions listed below. These changes may require that
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=== versions listed below. These changes may require that
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=== you modify your configuration files, dialplan or (in
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=== you modify your configuration files, dialplan or (in
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=== some cases) source code if you have your own Asterisk
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=== some cases) source code if you have your own Asterisk
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=== modules or patches. These files also includes advance
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=== modules or patches. These files also includes advance
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=== notice of any functionality that has been marked as
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=== notice of any functionality that has been marked as
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=== 'deprecated' and may be removed in a future release,
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=== 'deprecated' and may be removed in a future release,
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=== along with the suggested replacement functionality.
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=== along with the suggested replacement functionality.
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===
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===
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=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
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=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
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=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
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=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
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=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
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=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
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===
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===
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===========================================================
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===========================================================
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From 1.6.2 to 1.8:
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From 1.6.2 to 1.8:
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* Asterisk-addons no longer exists as an independent package.  Those modules
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* Asterisk-addons no longer exists as an independent package.  Those modules
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  now live in the addons directory of the main Asterisk source tree.  They
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  now live in the addons directory of the main Asterisk source tree.  They
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  are not enabled by default.  For more information about why modules live in
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  are not enabled by default.  For more information about why modules live in
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  addons, see README-addons.txt.
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  addons, see README-addons.txt.
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* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
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* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
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  users of this channel in the tree have been converted to LOG_NOTICE or removed
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  users of this channel in the tree have been converted to LOG_NOTICE or removed
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  (in cases where the same message was already generated to another channel).
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  (in cases where the same message was already generated to another channel).
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* The usage of RTP inside of Asterisk has now become modularized. This means
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* The usage of RTP inside of Asterisk has now become modularized. This means
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  the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
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  the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
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  If you are not using autoload=yes in modules.conf you will need to ensure
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  If you are not using autoload=yes in modules.conf you will need to ensure
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  it is set to load. If not, then any module which uses RTP (such as chan_sip)
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  it is set to load. If not, then any module which uses RTP (such as chan_sip)
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  will not be able to send or receive calls.
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  will not be able to send or receive calls.
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* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still 
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* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still 
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  remains. It now exists within app_chanspy.c and retains the exact same 
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  remains. It now exists within app_chanspy.c and retains the exact same 
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  functionality as before. 
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  functionality as before. 
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* The default behavior for Set, AGI, and pbx_realtime has been changed to implement
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* The default behavior for Set, AGI, and pbx_realtime has been changed to implement
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  1.6 behavior by default, if there is no [compat] section in asterisk.conf.  In
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  1.6 behavior by default, if there is no [compat] section in asterisk.conf.  In
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  prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.
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  prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.
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  Specifically, that means that pbx_realtime and res_agi expect you to use commas
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  Specifically, that means that pbx_realtime and res_agi expect you to use commas
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  to separate arguments in applications, and Set only takes a single pair of
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  to separate arguments in applications, and Set only takes a single pair of
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  a variable name/value.  The old 1.4 behavior may still be obtained by setting
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  a variable name/value.  The old 1.4 behavior may still be obtained by setting
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  app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of
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  app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of
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  asterisk.conf.
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  asterisk.conf.
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* The PRI channels in chan_dahdi can no longer change the channel name if a
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* The PRI channels in chan_dahdi can no longer change the channel name if a
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  different B channel is selected during call negotiation.  To prevent using
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  different B channel is selected during call negotiation.  To prevent using
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  the channel name to infer what B channel a call is using and to avoid name
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  the channel name to infer what B channel a call is using and to avoid name
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  collisions, the channel name format is changed.
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  collisions, the channel name format is changed.
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  The new channel naming for PRI channels is:
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  The new channel naming for PRI channels is:
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  DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
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  DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
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* The ChanIsAvail application has been changed so the AVAILSTATUS variable
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* The ChanIsAvail application has been changed so the AVAILSTATUS variable
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  no longer contains both the device state and cause code. The cause code
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  no longer contains both the device state and cause code. The cause code
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  is now available in the AVAILCAUSECODE variable. If existing dialplan logic
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  is now available in the AVAILCAUSECODE variable. If existing dialplan logic
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  is written to expect AVAILSTATUS to contain the cause code it needs to be
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  is written to expect AVAILSTATUS to contain the cause code it needs to be
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  changed to use AVAILCAUSECODE.
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  changed to use AVAILCAUSECODE.
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* ExternalIVR will now send Z events for invalid or missing files, T events
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* ExternalIVR will now send Z events for invalid or missing files, T events
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  now include the interrupted file and bugs in argument parsing have been
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  now include the interrupted file and bugs in argument parsing have been
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  fixed so there may be arguments specified in incorrect ways that were
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  fixed so there may be arguments specified in incorrect ways that were
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  working that will no longer work.
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  working that will no longer work.
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  Please see doc/externalivr.txt for details.
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  Please see doc/externalivr.txt for details.
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* OSP lookup application changes following variable names:
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* OSP lookup application changes following variable names:
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  OSPPEERIP to OSPINPEERIP
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  OSPPEERIP to OSPINPEERIP
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  OSPTECH to OSPOUTTECH
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  OSPTECH to OSPOUTTECH
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  OSPDEST to OSPDESTINATION
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  OSPDEST to OSPDESTINATION
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  OSPCALLING to OSPOUTCALLING
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  OSPCALLING to OSPOUTCALLING
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  OSPCALLED to OSPOUTCALLED
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  OSPCALLED to OSPOUTCALLED
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  OSPRESULTS to OSPDESTREMAILS
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  OSPRESULTS to OSPDESTREMAILS
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* The Manager event 'iax2 show peers' output has been updated.  It now has a
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* The Manager event 'iax2 show peers' output has been updated.  It now has a
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  similar output of 'sip show peers'.
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  similar output of 'sip show peers'.
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* VoiceMailMain and VMAuthenticate, if a '*' is entered in the first position

    
   
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  of a Mailbox or Password, will, if it exists, jump to the 'a' extension in

    
   
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  the current dialplan context.

    
   
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From 1.6.1 to 1.6.2:
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From 1.6.1 to 1.6.2:
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* SIP no longer sends the 183 progress message for early media by
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* SIP no longer sends the 183 progress message for early media by
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  default.  Applications requiring early media should use the
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  default.  Applications requiring early media should use the
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  progress() dialplan app to generate the progress message. 
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  progress() dialplan app to generate the progress message. 
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* The firmware for the IAXy has been removed from Asterisk.  It can be
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* The firmware for the IAXy has been removed from Asterisk.  It can be
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  downloaded from http://downloads.digium.com/pub/iaxy/.  To have Asterisk
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  downloaded from http://downloads.digium.com/pub/iaxy/.  To have Asterisk
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  install the firmware into its proper location, place the firmware in the
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  install the firmware into its proper location, place the firmware in the
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  contrib/firmware/iax/ directory in the Asterisk source tree before running
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  contrib/firmware/iax/ directory in the Asterisk source tree before running
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  "make install".
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  "make install".
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* T.38 FAX error correction mode can no longer be configured in udptl.conf;
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* T.38 FAX error correction mode can no longer be configured in udptl.conf;
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  instead, it is configured on a per-peer (or global) basis in sip.conf, with
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  instead, it is configured on a per-peer (or global) basis in sip.conf, with
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  the same default as was present in udptl.conf.sample.
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  the same default as was present in udptl.conf.sample.
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* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
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* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
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  instead, it is either supplied by the application servicing the T.38 channel
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  instead, it is either supplied by the application servicing the T.38 channel
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  (for a FAX send or receive) or calculated from the bridged endpoint's
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  (for a FAX send or receive) or calculated from the bridged endpoint's
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  maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
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  maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
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  allows for overriding the value supplied by a remote endpoint, which is useful
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  allows for overriding the value supplied by a remote endpoint, which is useful
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  when T.38 connections are made to gateways that supply incorrectly-calculated
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  when T.38 connections are made to gateways that supply incorrectly-calculated
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  maximum datagram sizes.
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  maximum datagram sizes.
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* There have been some changes to the IAX2 protocol to address the security
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* There have been some changes to the IAX2 protocol to address the security
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  concerns documented in the security advisory AST-2009-006.  Please see the
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  concerns documented in the security advisory AST-2009-006.  Please see the
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  IAX2 security document, doc/IAX2-security.pdf, for information regarding
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  IAX2 security document, doc/IAX2-security.pdf, for information regarding
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  backwards compatibility with versions of Asterisk that do not contain these
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  backwards compatibility with versions of Asterisk that do not contain these
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  changes to IAX2.
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  changes to IAX2.
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* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
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* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
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  has been renamed to 'directmedia', to better reflect what it actually does.
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  has been renamed to 'directmedia', to better reflect what it actually does.
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  In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
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  In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
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  starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
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  starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
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  option never had any effect on these cases, it only affected the re-INVITEs
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  option never had any effect on these cases, it only affected the re-INVITEs
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  used for direct media path setup. For MGCP and Skinny, the option was poorly
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  used for direct media path setup. For MGCP and Skinny, the option was poorly
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  named because those protocols don't even use INVITE messages at all. For
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  named because those protocols don't even use INVITE messages at all. For
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  backwards compatibility, the old option is still supported in both normal
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  backwards compatibility, the old option is still supported in both normal
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  and Realtime configuration files, but all of the sample configuration files,
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  and Realtime configuration files, but all of the sample configuration files,
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  Realtime/LDAP schemas, and other documentation refer to it using the new name.
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  Realtime/LDAP schemas, and other documentation refer to it using the new name.
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* The default console now will use colors according to the default background
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* The default console now will use colors according to the default background
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  color, instead of forcing the background color to black.  If you are using a
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  color, instead of forcing the background color to black.  If you are using a
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  light colored background for your console, you may wish to use the option
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  light colored background for your console, you may wish to use the option
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  flag '-W' to present better color choices for the various messages.  However,
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  flag '-W' to present better color choices for the various messages.  However,
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  if you'd prefer the old method of forcing colors to white text on a black
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  if you'd prefer the old method of forcing colors to white text on a black
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  background, the compatibility option -B is provided for this purpose.
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  background, the compatibility option -B is provided for this purpose.
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* SendImage() no longer hangs up the channel on transmission error or on
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* SendImage() no longer hangs up the channel on transmission error or on
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  any other error; in those cases, a FAILURE status is stored in
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  any other error; in those cases, a FAILURE status is stored in
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  SENDIMAGESTATUS and dialplan execution continues.  The possible
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  SENDIMAGESTATUS and dialplan execution continues.  The possible
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  return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and
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  return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and
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  UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'
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  UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'
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  has been replaced with 'UNSUPPORTED').  This change makes the
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  has been replaced with 'UNSUPPORTED').  This change makes the
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  SendImage application more consistent with other applications.
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  SendImage application more consistent with other applications.
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* skinny.conf now has separate sections for lines and devices.
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* skinny.conf now has separate sections for lines and devices.
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  Please have a look at configs/skinny.conf.sample and update
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  Please have a look at configs/skinny.conf.sample and update
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  your skinny.conf.
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  your skinny.conf.
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* Queue names previously were treated in a case-sensitive manner,
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* Queue names previously were treated in a case-sensitive manner,
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  meaning that queues with names like "sales" and "sALeS" would be
146
  meaning that queues with names like "sales" and "sALeS" would be
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  seen as unique queues. The parsing logic has changed to use
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  seen as unique queues. The parsing logic has changed to use
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  case-insensitive comparisons now when originally hashing based on
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  case-insensitive comparisons now when originally hashing based on
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  queue names, meaning that now the two queues mentioned as examples
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  queue names, meaning that now the two queues mentioned as examples
146
  earlier will be seen as having the same name.
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  earlier will be seen as having the same name.
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* The SPRINTF() dialplan function has been moved into its own module,
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* The SPRINTF() dialplan function has been moved into its own module,
149
  func_sprintf, and is no longer included in func_strings. If you use this
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  func_sprintf, and is no longer included in func_strings. If you use this
150
  function and do not use 'autoload=yes' in modules.conf, you will need
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  function and do not use 'autoload=yes' in modules.conf, you will need
151
  to explicitly load func_sprintf for it to be available.
155
  to explicitly load func_sprintf for it to be available.
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* The res_indications module has been removed.  Its functionality was important
157
* The res_indications module has been removed.  Its functionality was important
154
  enough that most of it has been moved into the Asterisk core.
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  enough that most of it has been moved into the Asterisk core.
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  Two applications previously provided by res_indications, PlayTones and
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  Two applications previously provided by res_indications, PlayTones and
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  StopPlayTones, have been moved into a new module, app_playtones.
160
  StopPlayTones, have been moved into a new module, app_playtones.
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161

   
158
* Support for Taiwanese was incorrectly supported with the "tw" language code.
162
* Support for Taiwanese was incorrectly supported with the "tw" language code.
159
  In reality, the "tw" language code is reserved for the Twi language, native
163
  In reality, the "tw" language code is reserved for the Twi language, native
160
  to Ghana.  If you were previously using the "tw" language code, you should
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  to Ghana.  If you were previously using the "tw" language code, you should
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  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
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  switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
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  specific localizations.  Additionally, "mx" should be changed to "es_MX",
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  specific localizations.  Additionally, "mx" should be changed to "es_MX",
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  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
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  Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
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  "cs", not "cz".
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  "cs", not "cz".
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166
* DAHDISendCallreroutingFacility() parameters are now comma-separated,
170
* DAHDISendCallreroutingFacility() parameters are now comma-separated,
167
  instead of the old pipe.
171
  instead of the old pipe.
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* res_jabber: autoprune has been disabled by default, to avoid misconfiguration 
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* res_jabber: autoprune has been disabled by default, to avoid misconfiguration 
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  that would end up being interpreted as a bug once Asterisk started removing 
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  that would end up being interpreted as a bug once Asterisk started removing 
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  the contacts from a user list.
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  the contacts from a user list.
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From 1.6.0.1 to 1.6.1:
177
From 1.6.0.1 to 1.6.1:
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175
* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
179
* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
176
  API calls were added in 1.6.0, so that modules that provide multiple
180
  API calls were added in 1.6.0, so that modules that provide multiple
177
  AGI commands could register/unregister them all with a single
181
  AGI commands could register/unregister them all with a single
178
  step. However, these API calls were not implemented properly, and did
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  step. However, these API calls were not implemented properly, and did
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  not allow the caller to know whether registration or unregistration
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  not allow the caller to know whether registration or unregistration
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  succeeded or failed. They have been redefined to now return success
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  succeeded or failed. They have been redefined to now return success
181
  or failure, but this means any code using these functions will need
185
  or failure, but this means any code using these functions will need
182
  be recompiled after upgrading to a version of Asterisk containing
186
  be recompiled after upgrading to a version of Asterisk containing
183
  these changes. In addition, the source code using these functions
187
  these changes. In addition, the source code using these functions
184
  should be reviewed to ensure it can properly react to failure
188
  should be reviewed to ensure it can properly react to failure
185
  of registration or unregistration of its API commands.
189
  of registration or unregistration of its API commands.
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187
* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
191
* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
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  to better match what it really does, and the argument order has been
192
  to better match what it really does, and the argument order has been
189
  changed to be consistent with other API calls that perform similar
193
  changed to be consistent with other API calls that perform similar
190
  operations.
194
  operations.
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From 1.6.0.x to 1.6.1:
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From 1.6.0.x to 1.6.1:
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197

   
194
* In previous versions of Asterisk, due to the way objects were arranged in
198
* In previous versions of Asterisk, due to the way objects were arranged in
195
  memory by chan_sip, the order of entries in sip.conf could be adjusted to
199
  memory by chan_sip, the order of entries in sip.conf could be adjusted to
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  control the behavior of matching against peers and users.  The way objects
200
  control the behavior of matching against peers and users.  The way objects
197
  are managed has been significantly changed for reasons involving performance
201
  are managed has been significantly changed for reasons involving performance
198
  and stability.  A side effect of these changes is that the order of entries
202
  and stability.  A side effect of these changes is that the order of entries
199
  in sip.conf can no longer be relied upon to control behavior.
203
  in sip.conf can no longer be relied upon to control behavior.
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204

   
201
* The following core commands dealing with dialplan have been deprecated: 'core
205
* The following core commands dealing with dialplan have been deprecated: 'core
202
  show globals', 'core set global' and 'core set chanvar'. Use the equivalent
206
  show globals', 'core set global' and 'core set chanvar'. Use the equivalent
203
  'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
207
  'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
204
  instead.
208
  instead.
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209

   
206
* In the dialplan expression parser, the logical value of spaces
210
* In the dialplan expression parser, the logical value of spaces
207
  immediately preceding a standalone 0 previously evaluated to
211
  immediately preceding a standalone 0 previously evaluated to
208
  true. It now evaluates to false.  This has confused a good many
212
  true. It now evaluates to false.  This has confused a good many
209
  people in the past (typically because they failed to realize the
213
  people in the past (typically because they failed to realize the
210
  space had any significance).  Since this violates the Principle of
214
  space had any significance).  Since this violates the Principle of
211
  Least Surprise, it has been changed.
215
  Least Surprise, it has been changed.
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216

   
213
* While app_directory has always relied on having a voicemail.conf or users.conf file
217
* While app_directory has always relied on having a voicemail.conf or users.conf file
214
  correctly set up, it now is dependent on app_voicemail being compiled as well.
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  correctly set up, it now is dependent on app_voicemail being compiled as well.
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* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
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* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
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  and you should start using that function instead for retrieving information about
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  and you should start using that function instead for retrieving information about
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  the channel in a technology-agnostic way.
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  the channel in a technology-agnostic way.
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* If you have any third party modules which use a config file variable whose
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* If you have any third party modules which use a config file variable whose
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  name ends in a '+', please note that the append capability added to this
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  name ends in a '+', please note that the append capability added to this
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  version may now conflict with that variable naming scheme.  An easy
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  version may now conflict with that variable naming scheme.  An easy
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  workaround is to ensure that a space occurs between the '+' and the '=',
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  workaround is to ensure that a space occurs between the '+' and the '=',
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  to differentiate your variable from the append operator.  This potential
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  to differentiate your variable from the append operator.  This potential
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  conflict is unlikely, but is documented here to be thorough.
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  conflict is unlikely, but is documented here to be thorough.
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* The "Join" event from app_queue now uses the CallerIDNum header instead of
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* The "Join" event from app_queue now uses the CallerIDNum header instead of
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  the CallerID header to indicate the CallerID number.
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  the CallerID header to indicate the CallerID number.
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* If you use ODBC storage for voicemail, there is a new field called "flag"
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* If you use ODBC storage for voicemail, there is a new field called "flag"
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  which should be a char(8) or larger.  This field specifies whether or not a
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  which should be a char(8) or larger.  This field specifies whether or not a
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  message has been designated to be "Urgent", "PRIORITY", or not.
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  message has been designated to be "Urgent", "PRIORITY", or not.
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/trunk/apps/app_voicemail.c
Revision 262004 New Change
 
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