Convert all release tags to Opsound music-on-hold.

For more details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/



git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/1.0.8@212958 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -1,50 +1,41 @@
- NOTE: Corrections or additions to the ChangeLog may be submitted to
-       http://bugs.digium.com.  Documentation and formatting fixes are not
-       not listed here.  A complete listing of changes is available through
-       the Asterisk-CVS mailing list hosted at http://lists.digium.com.
-
-Asterisk 1.0.9
-
- -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8
-
-Asterisk 1.0.8
+ NOTE: Corrections or additions to the ChangeLog may be submitted 
+       to http://bugs.digium.com.  Documentation and formatting
+       fixes are not listed here.  A complete listing of changes
+       is available through the Asterisk-CVS mailing list hosted
+       at http://lists.digium.com.
 
  -- chan_zap
-    -- Asterisk will now also look in the regular context for the fax extension
-       while executing a macro.  Previously, for this to work, the fax extension
-       would have to be included in the macro definition.
+    -- Asterisk will now also look in the regular context for the fax extension while
+       executing a macro.  Previously, for this to work, the fax extension would have
+       to be included in the macro definition.
     -- On some systems, ALERTING will be sent after PROCEEDING, so code has been
        added to account for this case.
-    -- If no extension is specified on an overlap call, the 's' extension will 
-       be used.
+    -- If no extension is specified on an overlap call, the 's' extension will be used.
  -- chan_sip
-    -- We no longer send a "to" tag on "100 Trying" messages, as it is 
-       inappropriate to do so.
-    -- We now respond correctly to an invite for T.38 with a "488 Not acceptable
-       here"
-    -- We now discard saved tags on 401/407 responses in case the provider we're
-       talking to tries to pull a dirty trick on us and change it.
-    -- rtptimeout options will now be correctly set on a peer basis rather than
-       only global
+    -- We no longer send a "to" tag on "100 Trying" messages, as it is inappropriate
+       to do so.
+    -- We now respond correctly to an invite for T.38 with a "488 Not acceptable here"
+    -- We now discard saved tags on 401/407 responses in case the provider we're talking
+       to tries to pull a dirty trick on us and change it.
+    -- rtptimeout options will now be correctly set on a peer basis rather than only global
  -- chan_mgcp
     -- Fixed setting of accountcode
     -- Fixed where *67 to block callerid only worked for first call
  -- chan_agent
-    -- We now will not pass audio until the agent has acked the call if the 
-       configuration
+    -- We now will not pass audio until the agent has acked the call if the configuration
        is set up for the agent to do so.
  -- chan_alsa
     -- Fixed problems with the unloading of this module
  -- res_agi
-    -- A fix has been added to prevent calls from being hung up when more than 
-       one call is executing an AGI script calling the GET DATA command.
-    -- AGI scripts will now continue to run even if a file was not found with
-       the GET DATA command.
-    -- When calling SAY NUMBER with a number like 09, we will now say "nine" 
-       instead of "zero"
+    -- A fix has been added to prevent calls from being hung up when more than one
+       call is executing an AGI script calling the GET DATA command.
+    -- AGI scripts will now continue to run even if a file was not found with the
+       GET DATA command.
+    -- When calling SAY NUMBER with a number like 09, we will now say "nine" instead
+       of "zero"
  -- app_dial
-    -- There was a problem where text frames would not be forwarded before the
-       channel has been answered. 
+    -- There was a problem where text frames would not be forwarded before the channel
+       has been answered. 
  -- app_disa
     -- Fixed the timeout used when no password is set
  -- app_queue
@@ -54,17 +45,15 @@ Asterisk 1.0.8
  -- say.c
     -- A problem has been fixed with saying the date in Spanish.
  -- Makefile
-    -- A line was missing for the autosupport script that caused "make rpm" to 
-       fail
+    -- A line was missing for the autosupport script that caused "make rpm" to fail
  -- format_wav_gsm
-    -- Fixed a problem with wav formatting that prevented files from being 
-       played in some media players
+    -- Fixed a problem with wav formatting that prevented files from being played
+       in some media players
  -- pbx_spool
-    -- Fixed if the last line of text in a file for the call spool did not 
-       contain a new line, it would not be processed
+    -- Fixed if the last line of text in a file for the call spool did not contain
+       a new line, it would not be processed
  -- logger
-    -- Fixed the logger so that color escape sequences wouldn't be sent to the 
-       logs
+    -- Fixed the logger so that color escape sequences wouldn't be sent to the logs
  -- format_sln
     -- A lot of changes were made to correctly handle signed linear format on
        big endian machines
@@ -74,91 +63,79 @@ Asterisk 1.0.8
 Asterisk 1.0.7
 
  -- chan_sip
-    -- The fix for some codec availibility issues in 1.0.6 caused music on hold
-       problems, but has now been fixed.
+    -- The fix for some codec availibility issues in 1.0.6 caused music on hold problems,
+       but has now been fixed.
  -- chan_skinny
     -- A check has been added to avoid a crash.
  -- chan_iax2
-    -- A feature has been added to CVS head to have the option of sending 
-       timestamps with trunk frames.  It is not supported in 1.0, but a change 
-       has been made so that it will at least not choke if sent trunk
-       timestamps.
+    -- A feature has been added to CVS head to have the option of sending timestamps with
+       trunk frames.  It is not supported in 1.0, but a change has been made so that it
+       will at least not choke if sent trunk timestamps.
  -- app_voicemail
     -- Some checks have been added to avoid a crash.
  -- speex
-    -- The path /usr/include/speex has been added for a place to look for the 
-       speex header.
+    -- The path /usr/include/speex has been added for a place to look for the speex header.
 
 Asterisk 1.0.6
 
  -- chan_iax2:
     -- Fixed a bug dealing with a division by zero that could cause a crash
  -- chan_sip:
-    -- Behavior was changed so that when a registration fails due to DNS 
-       resolution issues, a retry will be attempted in 20 seconds.
-    -- Peer settings were not reset to null values when reloading the 
-       configuration file. Behavior has been changed so that these values are 
-       now cleared.
+    -- Behavior was changed so that when a registration fails due to DNS resolution issues,
+       a retry will be attempted in 20 seconds.
+    -- Peer settings were not reset to null values when reloading the configuration file.
+       Behavior has been changed so that these values are now cleared.
     -- 'restrictcid' now properly works on MySQL peers.
     -- Only use the default callerid if it has been specified.
-    -- Asterisk was not sending the same From: line in SIP messages during 
-       certain times. Fixed to make sure it stays the same. This makes some 
-       providers happier, to a working state.
-    -- Certain circumstances involving a blank callerid caused asterisk to 
-       segmentation fault.
-    -- There was a problem incorrectly matching codec availablity when global 
-       preferences were different from that of the user.  To fix this, 
-       processing of SDP data has been moved to after determining who the call 
-       is coming from.
-    -- Asterisk would run out of RTP ports while waiting for SUBSCRIBE's to 
-       expire even though an RTP port isn't needed in this case.  This has been
-       fixed by releasing the ports early.
+    -- Asterisk was not sending the same From: line in SIP messages during certain times.
+       Fixed to make sure it stays the same. This makes some providers happier, to a working state.
+    -- Certain circumstances involving a blank callerid caused asterisk to segmentation fault.
+    -- There was a problem incorrectly matching codec availablity when global preferences were 
+       different from that of the user.  To fix this, processing of SDP data has been moved 
+       to after determining who the call is coming from.
+    -- Asterisk would run out of RTP ports while waiting for SUBSCRIBE's to expire even though
+       an RTP port isn't needed in this case.  This has been fixed by releasing the ports early.
  -- chan_zap:
-    -- During a certain scenario when using flash and '#' transfers you would 
-       hear the other person and the music they were hearing. This has been 
-       fixed.
+    -- During a certain scenario when using flash and '#' transfers you would hear the
+       other person and the music they were hearing. This has been fixed.
     -- A fix for a compilation issue with gcc4 was added.
  -- chan_modem_bestdata:
     -- A fix for a compilation issue with gcc4 was added.
  -- format_g729:
-    -- Treat a 10-byte read as an end of file indication instead of an error. 
-       Some G729 encoders like to put 10-bytes at the end to indicate this.
+    -- Treat a 10-byte read as an end of file indication instead of an error. Some G729 encoders
+       like to put 10-bytes at the end to indicate this.
  -- res_features:
-    -- During certain situations when parking a call, both endpoints would get 
-       musiconhold. This has been fixed so the individual who parked the call 
-       will hear the digits and not musiconhold.
+    -- During certain situations when parking a call, both endpoints would get musiconhold.
+       This has been fixed so the individual who parked the call will hear the digits and not
+       musiconhold.
  -- app_dial:
-    -- DIALEDPEERNUMBER is now being set, so if you attempted to use it in the 
-       past and failed, it should work now.
-    -- A callerid change caused many headaches, this has been reversed to the 
-       original 1.0 behavior.
-    -- A crash caused with the combination of the 'g' option and # transfer was
-       fixed.
+    -- DIALEDPEERNUMBER is now being set, so if you attempted to use it in the past and failed
+       it should work now.
+    -- A callerid change caused many headaches, this has been reversed to the original 1.0 behavior.
+    -- A crash caused with the combination of the 'g' option and # transfer was fixed.
  -- app_voicemail:
-    -- If two people hit the voicemail system at the same time, and were leaving
-       a message the second message was overwriting the first. This has been 
-       fixed so that each one is distinct and will not overwrite eachother.
+    -- If two people hit the voicemail system at the same time, and were leaving a message
+       the second message was overwriting the first. This has been fixed so that each one
+       is distinct and will not overwrite eachother.
  -- cdr_tds:
-    -- If the server you were using was going down, it had the potential to 
-       bring your asterisk server down with it. Extra stuff has been added so 
-       as to bring in more error/connection checking.
+    -- If the server you were using was going down, it had the potential to bring your asterisk
+       server down with it. Extra stuff has been added so as to bring in more error/connection
+       checking.
  -- cdr_pgsql:
     -- This will now attempt to reconnect after a connection problem.
  -- IAXY firmware:
-    -- This has been updated to version 23.  It includes a fix for lost
-       registrations.
+    -- This has been updated to version 23.  It includes a fix for lost registrations.
  -- internals
-    -- Behavior was changed for 'show codec <number>' to make it more intuitive.
-    -- DNS failures and asterisk do not get along too well, this is not totally
-       the case anymore.
-    -- Asterisk will now handle DNS failures at startup more gracefully, and 
-       won't crash and burn
-    -- Choosing to append to a wave file would render the outputted wave file 
-       corrupt. Appending now works again.
-    -- If you failed to define certain keys, asterisk had the potential to crash
-       when seeing if you had used them.
-    -- Attempting to use such things as ${EXTEN:-1} gave a wrong return value. 
-       However, this was never a documented feature...
+    -- Behavior was changed for 'show codec <number>' to make it more intuitive. (kshumard)
+    -- DNS failures and asterisk do not get along too well, this is not totally the case anymore.
+    -- Asterisk will now handle DNS failures at startup more gracefully, and won't crash and
+       burn.
+    -- Choosing to append to a wave file would render the outputted wave file corrupt. Appending
+       now works again.
+    -- If you failed to define certain keys, asterisk had the potential to crash when seeing if you had
+       used them.
+    -- Attempting to use such things as ${EXTEN:-1} gave a wrong return value. However, this was never
+       a documented feature...
 
 Asterisk 1.0.5
 

pbx/pbx_config.c

@@ -1687,10 +1687,15 @@ static int pbx_load_module(void)
 								else
 									data = "";
 							}
-							pbx_substitute_variables_helper(NULL, ext, realext, sizeof(realext)-1);
-							stringp = realext;
-							ext = strsep(&stringp, "/");
-							cidmatch = stringp;
+							pbx_substitute_variables_helper(NULL, ext, realext, sizeof(realext)-1);	
+							cidmatch = strchr(ext, '/');
+							if (cidmatch) {
+								*cidmatch = '\0';
+								cidmatch++;
+							}
+							stringp=ext;
+							strsep(&stringp, "/");
+
 							if (!data)
 								data="";
 							while(*appl && (*appl < 33)) appl++;