This commit was manufactured by cvs2svn to create tag 'v1-0-8'.

git-svn-id: http://svn.digium.com/svn/asterisk/tags/v1-0-8@5968 f38db490-d61c-443f-a65b-d21fe96a405b

14 files changed, 265 insertions, 47 deletions

diff --git a/configs/agents.conf.sample b/configs/agents.conf.sample
index bde308134..d8641e60b 100755
--- a/configs/agents.conf.sample
+++ b/configs/agents.conf.sample
@@ -11,7 +11,7 @@
 ;autologoff=15
 ;
 ; Define ackcall to require an acknowledgement by '#' when
-; an agent logs in over agentcallpark.  Default is "yes".
+; an agent logs in using agentcallbacklogin.  Default is "yes".
 ;
 ;ackcall=yes
 ;
@@ -32,7 +32,7 @@
 ;
 ;updatecdr=no
 ;
-; Group memberships for agents (may change in mid-file just)
+; Group memberships for agents (may change in mid-file)
 ; 
 ;group=3
 ;group=1,2
diff --git a/configs/enum.conf.sample b/configs/enum.conf.sample
index 905f10fe1..1e0c1e1be 100755
--- a/configs/enum.conf.sample
+++ b/configs/enum.conf.sample
@@ -2,6 +2,7 @@
 ; ENUM Configuration for resolving phone numbers over DNS
 ;
 ; Sample config for Asterisk
+; This file is reloaded at "reload" in the CLI
 ;
 [general]
 ;
diff --git a/configs/extensions.conf.sample b/configs/extensions.conf.sample
index 624aff8aa..ed8785a7b 100755
--- a/configs/extensions.conf.sample
+++ b/configs/extensions.conf.sample
@@ -3,6 +3,9 @@
 ; the pbx_config module. This is where you configure all your 
 ; inbound and outbound calls in Asterisk. 
 ; 
+; This configuration file is reloaded 
+; - With the "extensions reload" command in the CLI
+; - With the "reload" command (that reloads everything) in the CLI
 
 ;
 ; The "General" category is for certain variables.  
@@ -38,6 +41,16 @@ CONSOLE=Console/dsp				; Console interface for demo
 IAXINFO=guest					; IAXtel username/password
 ;IAXINFO=myuser:mypass
 TRUNK=Zap/g2					; Trunk interface
+;
+; Note the 'g2' in the TRUNK variable above. It specifies which group (defined
+; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in
+; the specified group. The four possible options are:
+;
+; g: select the lowest-numbered non-busy Zap channel (aka. ascending sequential hunt group).
+; G: select the highest-numbered non-busy Zap channel (aka. descending sequential hunt group).
+; r: use a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group).
+; R: use a round-robin search, starting at the next lowest channel than last time (aka. descending rotary hunt group).
+;
 TRUNKMSD=1					; MSD digits to strip (usually 1 or 0)
 ;TRUNK=IAX2/user:pass@provider
 
diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample
index 98ffd3d9c..d34d9bc87 100755
--- a/configs/iax.conf.sample
+++ b/configs/iax.conf.sample
@@ -1,6 +1,9 @@
 
 ; Inter-Asterisk eXchange driver definition
 ;
+; This configuration is re-read at reload
+; or with the CLI command
+; 	reload chan_iax2.so
 ;
 ; General settings, like port number to bind to, and
 ; an option address (the default is to bind to all
diff --git a/configs/indications.conf.sample b/configs/indications.conf.sample
index df05da4ce..7298a3612 100755
--- a/configs/indications.conf.sample
+++ b/configs/indications.conf.sample
@@ -1,14 +1,14 @@
+; indications.conf
+; Configuration file for location specific tone indications
+; used by the pbx_indications module.
 ;
-; Static indications configuration files, used by
-; the pbx_indications module.
+; The [general] category is for certain global variables.
+; All other categories are interpreted as location specific indications
 ;
-; The "general" category is for certain variables.  All other categories
-; are interpreted as indication countries
-;
-; Please note that there are NOT spaces allowed in lists!
 ;
 [general]
-country=us
+country=us		; default location
+
 
 ; [example]
 ; description = string
@@ -23,28 +23,36 @@ country=us
 ; busy = tonelist
 ;      Set of tones played when the receiving end is busy.
 ; congestion = tonelist
-;      Set of tones played when the is some congestion (on the network?)
+;      Set of tones played when there is some congestion (on the network?)
 ; callwaiting = tonelist
-;      Set of tones played when there is a callwaiting in the background.
+;      Set of tones played when there is a call waiting in the background.
 ; dialrecall = tonelist
-;      Set of tones played when there is somebody ???
+;      Not well defined, many phone systems play a recall dial tone after hook flash
 ; record = tonelist
-;      Set of tones played whenver we feel like it ???
+;      Set of tones played when call recording is in progress
 ; info = tonelist
-;      Set of tones played when information is to be tranfered?
-; every other variable will be available as a shortcut for the "PlayList" command
-; but will not automaticly be used by Asterisk.
+;      Set of tones played with special information messages (e.g., "number is out of service")
+; 'name' = tonelist
+;	Every other variable will be available as a shortcut for the "PlayList" command
+;	but will not automaticly be used by Asterisk.
+;
 ;
-; The tonelist itself is defined by a sequence of elements, seperated by ,'s.
-; Each element consist of a frequency (f) with a possible frequency attached
-; (f1+f2) to it. Behind the frequency there is an optional duration, in
-; milliseconds. If the element starts with a !, that element is NOT repeat,
-; so only if all elements start with !, the tonelist is time-limited, all
-; others will repeat indefinitly.
+; The tonelist itself is defined by a comma-separated sequence of elements.
+; Each element consist of a frequency (f) with an optional duration (in ms)
+; attached to it (f/duration). The frequency component may be a mixture of two
+; frequencies (f1+f2) or a frequency modulated by another frequency (f1*f2).
+; The implicit modulation depth is fixed at 90%, though.
+; If the list element starts with a !, that element is NOT repeated,
+; therefore, only if all elements start with !, the tonelist is time-limited,
+; all others will repeat indefinitely.
 ;
-; In tech-talk:
+; concisely:
+;   element = [!]freq[+|*freq2][/duration]
 ;   tonelist = element[,element]*
-;   element = [!]freq[+freq2][/duration]
+;
+; Please note that SPACES ARE NOT ALLOWED in tone lists!
+;
+
 
 ; Tone definitions for Chile
 ; According to specs from Telefonica CTC Chile
@@ -61,7 +69,6 @@ record = 1400/500,0/15000
 info = 950/333,1400/333,1800/333,0/1000
 
 [tw]
-; 參考資料 :
 ; http://nemesis.lonestar.org/reference/telecom/signaling/dialtone.html
 ; http://nemesis.lonestar.org/reference/telecom/signaling/busy.html
 ; http://www.iproducts.com.tw/ee/kylink/06ky-1000a.htm
@@ -69,14 +76,11 @@ info = 950/333,1400/333,1800/333,0/1000
 ; http://www.nettwerked.net/tones.txt
 ; http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/taiw_sup/taiw2.htm
 ;
-; 詢問中華電信後得到的回覆 :
-; dial tone 350+440Hz continuous
 ; busy tone 480+620Hz 0.5 sec. on ,0.5 sec. off
 ; reorder tone 480+620Hz 0.25 sec. on,0.25 sec. off
 ; ringing tone 440+480Hz 1 sec. on ,2 sec. off
 ;
 description = Taiwan
-alias = tw
 ringcadance = 1000,4000
 dial = 350+440
 busy = 480+620/500,0/500
@@ -181,15 +185,36 @@ info = 950/330,1400/330,1800/330,0/1000
 [uk]
 description = United Kingdom
 ringcadance = 400,200,400,2000
+; These are the official tones taken from BT SIN350. The actual tones
+; used by BT include some volume differences so sound slightly different
+; from Asterisk-generated ones.
 dial = 350+440
+; Special dial is the intermittent dial tone heard when, for example,
+; you have a divert active on the line
+specialdial = 350+440/750,440/750
+; Busy is also called "Engaged"
 busy = 400/375,0/375
-ring = 400+450/400,0/200,400+450/400,0/2000
+; "Congestion" is the Beep-bip engaged tone
 congestion = 400/400,0/350,400/225,0/525
-callwaiting = 440/100,0/4000
-dialrecall = 350+440
-; XXX Not sure about the RECORDTONE
-record = 1400/500,0/10000
-info = 950/330,1400/330,1800/330
+; "Special Congestion" is not used by BT very often if at all
+specialcongestion = 400/200,1004/300
+unobtainable = 400
+ring = 400+450/400,0/200,400+450/400,0/2000
+callwaiting = 400/100,0/4000
+; BT seem to use "Special Call Waiting" rather than just "Call Waiting" tones
+specialcallwaiting = 400/250,0/250,400/250,0/250,400/250,0/5000
+; "Pips" used by BT on payphones. (Souds wrong, but this is what BT claim it
+; is and I've not used a payphone for years)
+creditexpired = 400/125,0/125
+; These two are used to confirm/reject service requests on exchanges that
+; don't do voice announcements.
+confirm = 1400
+switching = 400/200,0/400,400/2000,0/400
+; This is the three rising tones Doo-dah-dee "Special Information Tone",
+; usually followed by the BT woman saying an appropriate message.
+info = 950/330,0/15,1400/330,0/15,1800/330,0/1000
+; Not listed in SIN350
+record = 1400/500,0/60000
 
 [fi]
 description = Finland
@@ -317,3 +342,120 @@ callwaiting = !400/200,!0/3000,!400/200,!0/3000,!400/200,!0/3000,!400/200
 dialrecall = !400/100!0/100,!400/100,!0/100,!400/100,!0/100,400
 record = 1400/425,0/15000
 info = 400/750,0/100,400/750,0/100,400/750,0/100,400/750,0/400
+
+; Singapore
+; Reference: http://www.ida.gov.sg/idaweb/doc/download/I397/ida_ts_pstn1_i4r2.pdf 
+; Frequency specs are:   425 Hz +/- 20Hz; 24 Hz +/- 2Hz; modulation depth 100%; SIT +/- 50Hz
+[sg]
+description = Singapore
+ringcadence = 400,200,400,2000
+dial       = 425
+ring       = 425*24/400,0/200,425*24/400,0/2000	; modulation should be 100%, not 90%
+busy       = 425/750,0/750
+congestion = 425/250,0/250
+callwaiting= 425*24/300,0/200,425*24/300,0/3200
+stutter    = !425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,!425/200,!0/200,!425/600,!0/200,425
+info       = 950/330,1400/330,1800/330,0/1000	; not currently in use acc. to reference
+dialrecall = 425*24/500,0/500,425/500,0/2500	; unspecified in IDA reference, use repeating Holding Tone A,B
+record     = 1400/500,0/15000			; unspecified in IDA reference, use 0.5s tone every 15s
+; additionally defined in reference
+nutone     = 425/2500,0/500
+intrusion  = 425/250,0/2000
+warning    = 425/624,0/4376			; end of period tone, warning
+acceptance = 425/125,0/125
+holdinga   = !425*24/500,!0/500			; followed by holdingb
+holdingb   = !425/500,!0/2500
+
+[hu]
+description = Hungary
+ringcadance = 1000,4000
+dial = 425
+busy = 425/300,0/300
+ring = 425/1250,0/3750
+congestion = 425/300,0/300
+callwaiting = 425/40,0/1960
+; Dialrecall not used in Hungary, use standard (using UK standard)
+dialrecall = 350+440
+; Record tone is not used in Hungary, use busy tone
+record = 425/250,0/250
+info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
+
+[lt]
+description = Lithuania
+ringcadence = 1000,4000
+dial = 425
+busy = 425/350,0/350
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+callwaiting = 425/150,0/150,425/150,0/4000
+; XXX I'm making up the dialrecall tone XXX
+dialrecall = 425/500,0/50
+; XXX I'm making up the record tone XXX
+record = 1400/500,0/15000
+info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
+
+[pl]
+description = Poland
+ringcadence = 1000,4000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/4000
+congestion = 425/500,0/500
+callwaiting = 425/150,0/150,425/150,0/4000
+; DIALRECALL - not specified
+dialrecall = 425/500,0/50
+; RECORDTONE - not specified
+record = 1400/500,0/15000
+; 950/1400/1800 3x0.33 on 1.0 off  repeated 3 times 
+info = !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000
+
+
+[pt]
+description = Portugal
+ringcadance = 1000,5000
+dial = 425
+busy = 425/500,0/500
+ring = 425/1000,0/5000
+congestion = 425/200,0/200
+callwaiting = 440/300,0/10000
+dialrecall = 425/1000,0/200
+record = 1400/500,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+
+[ee]
+description = Estonia
+ringcadance = 1000,4000
+dial = 425
+busy = 425/300,0/300
+ring = 425/1000,0/4000
+congestion = 425/200,0/200
+callwaiting = 950/650,0/325,950/325,0/30,1400/1300,0/2600
+dialrecall = 425/650,0/25
+record = 1400/500,0/15000
+info = 950/650,0/325,950/325,0/30,1400/1300,0/2600
+
+[mx]
+description = Mexico
+ringcadance = 2000,4000
+dial = 425
+busy = 425/250,0/250
+ring = 425/1000,0/4000
+congestion = 425/250,0/250
+callwaiting = 425/200,0/600,425/200,0/10000
+dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+record = 1400/500,0/15000
+info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
+
+[se]
+description = Sweden
+ringcadance = 1000,5000
+dial = 425
+busy = 425/250,0/250
+ring = 425/1000,0/5000
+congestion = 425/250,0/750
+callwaiting = 425/200,0/500,425/200,0/9100
+dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
+record = 1400/500,0/15000
+info = !950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,!0/2024,!950/332,!0/24,!1400/332,!0/24,!1800/332,0
+stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
+; stutter = 425/320,0/20        ; Real swedish standard, not used for now
diff --git a/configs/logger.conf.sample b/configs/logger.conf.sample
index bf28f2f40..338f2bb1f 100755
--- a/configs/logger.conf.sample
+++ b/configs/logger.conf.sample
@@ -3,6 +3,9 @@
 ;
 ; In this file, you configure logging to files or to
 ; the syslog system.
+;
+; "logger reload" at the CLI will reload configuration
+; of the logging system.
 
 [general]
 ; Customize the display of debug message time stamps
@@ -40,10 +43,6 @@ console => notice,warning,error
 messages => notice,warning,error
 ;full => notice,warning,error,debug,verbose
 ;
-; Uncomment the following line (with *no* events) if you want the
-; queue log to automatically rotate.
-;
-;queue_log =>
 
 ;syslog keyword : This special keyword logs to syslog facility 
 ;
diff --git a/configs/manager.conf.sample b/configs/manager.conf.sample
index 4560904e2..479c91938 100755
--- a/configs/manager.conf.sample
+++ b/configs/manager.conf.sample
@@ -1,5 +1,23 @@
 ;
-; Asterisk Call Management support
+; AMI - The Asterisk Manager Interface
+; 
+; Third party application call management support
+; and PBX event supervision
+;
+; This configuration file is read every time someone
+; logs in
+;
+; Use the "show manager commands" at the CLI to list
+; availabale manager commands and their authorization
+; levels.
+;
+; "show manager command <command>" will show a help text.
+;
+; ------------------- SECURITY NOTE -----------------
+; Note that you should not enable the AMI on a public
+; IP address. If needed, block this TCP port with
+; iptables (or another FW software) and reach it
+; with IPsec, SSH or SSL vpn tunnel
 ;
 [general]
 enabled = no
@@ -10,5 +28,7 @@ bindaddr = 0.0.0.0
 ;secret = mysecret
 ;deny=0.0.0.0/0.0.0.0
 ;permit=209.16.236.73/255.255.255.0
+;
+; Authorization for various classes 
 ;read = system,call,log,verbose,command,agent,user
 ;write = system,call,log,verbose,command,agent,user
diff --git a/configs/meetme.conf.sample b/configs/meetme.conf.sample
index c78cc21ac..417ce862e 100755
--- a/configs/meetme.conf.sample
+++ b/configs/meetme.conf.sample
@@ -2,6 +2,8 @@
 ; Configuration file for MeetMe simple conference rooms
 ; for Asterisk of course.
 ;
+; This configuration file is read every time you call app meetme()
+;
 [rooms]
 ;
 ; Usage is conf => confno[,pin]
diff --git a/configs/modem.conf.sample b/configs/modem.conf.sample
index 3af4530b0..bca405d96 100755
--- a/configs/modem.conf.sample
+++ b/configs/modem.conf.sample
@@ -84,7 +84,7 @@ mode=immediate
 ; two other devices, which are in group '1' and are used when an
 ; outgoing dial used: exten => s,1,Dial,Modem/g1:1234|60|r
 ; (we do not need more outgoing devices, since ISDN2 has only 2 channels.)
-; Lines can be in more than one group (1-31); comma seperated list.
+; Lines can be in more than one group (1-31); comma separated list.
 ;
 group=1		; group=1,2,3,9-12
 ;msn=50780023
diff --git a/configs/queues.conf.sample b/configs/queues.conf.sample
index 7ba488ad3..a0e73e04d 100755
--- a/configs/queues.conf.sample
+++ b/configs/queues.conf.sample
@@ -17,10 +17,10 @@
 ;
 ; A sample call queue
 ;
-; Music sets which musiconhold applies for this particular
-; call queue
+; Musiconhold sets which music applies for this particular
+; call queue (configure classes in musiconhold.conf)
 ;
-;music = default
+;musiconhold = default
 ;
 ; An announcement may be specified which is played for the member as
 ; soon as they answer a call, typically to indicate to them which queue
@@ -92,6 +92,7 @@
 ;queue-minutes = queue-minutes			;	("minutes.")
 ;queue-seconds = queue-seconds			;	("seconds.")
 ;queue-thankyou = queue-thankyou		;	("Thank you for your patience.")
+;queue-lessthan = queue-less-than		;       ("less than")
 ;
 ; Calls may be recorded using Asterisk's monitor resource
 ; This can be enabled from within the Queue application, starting recording
@@ -109,10 +110,16 @@
 ;
 ; monitor-join = yes
 ;
-; If you wish to allow queues that have no members currently to be joined, set this to yes
+; If you wish to allow queues that have no members currently to be joined, then set this
+; to yes.  Note that this is for use with dynamic queue members!
 ;
 ; joinempty = yes
 ;
+; If you wish to remove callers from the queue if there are no agents present, then set
+; this to yes.  Note that this is for use with dynamic queue members!
+;
+; leavewhenempty = yes
+;
 ; Asterisk can generate AgentCalled events when an agent is rung, if this is turned on
 ; (may generate a LOT of extra manager events)
 ;
@@ -128,6 +135,12 @@
 ;member => Agent/1001
 ;member => Agent/1002
 
+;
+; Note that using agent groups is probably not what you want.  Strategies do
+; not propagate down to the Agent system so if you want round robin, least
+; recent, etc, you should list all the agents in this file individually and not
+; use agent groups.
+;
 ;member => Agent/@1		; Any agent in group 1
 ;member => Agent/:1,1		; Any agent in group 1, wait for first
                                 ; available, but consider with penalty
diff --git a/configs/rtp.conf.sample b/configs/rtp.conf.sample
index 76327cd95..a9a455c92 100755
--- a/configs/rtp.conf.sample
+++ b/configs/rtp.conf.sample
@@ -5,6 +5,8 @@
 ;
 ; RTP start and RTP end configure start and end addresses
 ;
+; Defaults are rtpstart=5000 and rtpend=31000
+;
 rtpstart=10000
 rtpend=20000
 ;
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index de46d9439..5b9053de5 100755
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -19,6 +19,9 @@
 ;
 ;   sip debug			Show all SIP messages
 ;
+;   reload chan_sip.so		Reload configuration file
+;				Active SIP peers will not be reconfigured
+;
 
 [general]
 context=default			; Default context for incoming calls
@@ -131,7 +134,6 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ; context                     context
 ; permit                      permit
 ; deny                        deny
-; auth                        auth
 ; secret                      secret
 ; md5secret                   md5secret
 ; dtmfmode                    dtmfmode
@@ -229,6 +231,18 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 ;mailbox=1234@context,2345      ; Mailbox(-es) for message waiting indicator
 
 
+;[polycom]
+;type=friend			; Friends place calls and receive calls
+;context=from-sip		; Context for incoming calls from this user
+;secret=blahpoly
+;host=dynamic			; This peer register with us
+;dtmfmode=rfc2833		; Choices are inband, rfc2833, or info
+;username=polly			; Username to use in INVITE until peer registers
+;disallow=all
+;allow=ulaw                     ; dtmfmode=inband only works with ulaw or alaw!
+;progressinband=no		; Polycom phones don't work properly with "never"
+
+
 ;[pingtel]
 ;type=friend
 ;username=pingtel
diff --git a/configs/voicemail.conf.sample b/configs/voicemail.conf.sample
index 315611a76..5b55eed09 100755
--- a/configs/voicemail.conf.sample
+++ b/configs/voicemail.conf.sample
@@ -55,8 +55,10 @@ maxlogins=3
 ;
 ; Note: The emailbody config row can be up to 512 characters due to a limitation in 
 ;       asterisk config files.
-;emailsubject=New VM (${VM_MSGNUM}) - ${VM_DUR} long in mailbox ${VM_MAILBOX} from ${VM_CALLERID} 
-;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE} so you might\nwant to check it when you get a chance.  Thanks!\n\n\t\t\t\t--Asterisk\n
+;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
+; The following definition is very close to the default, but the default shows just 
+; the CIDNAME, if it is not null, else just the CIDNUM, or "an unknown caller" if they are both null.
+;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance.  Thanks!\n\n\t\t\t\t--Asterisk\n
 
 ;
 ; You can override the default program to send e-mail if you wish, too
diff --git a/configs/zapata.conf.sample b/configs/zapata.conf.sample
index c692a27b8..b7b660318 100755
--- a/configs/zapata.conf.sample
+++ b/configs/zapata.conf.sample
@@ -177,6 +177,13 @@ callwaiting=yes
 ;
 usecallingpres=yes
 ;
+; Some countries (UK) have ring tones with different ring tones (ring-ring),
+; which means the callerid needs to be set later on, and not just after
+; the first ring, as per the default. 
+;
+;sendcalleridafter=1
+;
+;
 ; Support Caller*ID on Call Waiting
 ;
 callwaitingcallerid=yes