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# openssl genrsa -out ca.key 4096
# openssl req -new -x509 -days 3650 -key ca.key -out ca.crt
# openssl genrsa -des3 -out pbx.key 4096
# openssl req -new -key argus.key -out pbx.csr
# openssl x509 -req -days 3650 -in pbx.csr -CA ca.crt -CAkey ca.key -set_serial 01 -out pbx.crt
# openssl rsa -in pbx.key -out pbx.key.insecure
# cat pbx.crt > pbx.pem
# cat pbx.key.insecure >> pbx.pem
# mv pbx.pem ca.crt /var/lib/asterisk/
# restorecon -R /var/lib/asterisk/
# chown asterisk:asterisk /var/lib/asterisk

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; --- Asterisk's Primary Configuration ---
; Gentlemen

[directories](!)
  astetcdir => /etc/asterisk                ; The Asterisk configuration files
  astmoddir => /usr/lib64/asterisk/modules  ; The loadable modules
  astvarlibdir => /usr/share/asterisk       ; The base location for variable state information used by
                                            ; various parts of Asterisk. This includes items that are
                                            ; written out by Asterisk at runtime
  astdbdir => /var/spool/asterisk           ; Asterisk will store its internal database in this directory
                                            ; as a file called astdb
  astkeydir => /var/lib/asterisk            ; Asterisk will use a subdirectory called keys in this
                                            ; directory as the default location for loading keys for
                                            ; encryption
  astdatadir => /usr/share/asterisk         ; This is the base directory for system-provided data, such as
                                            ; the sound files that come with Asterisk
  astagidir => /usr/share/asterisk/agi-bin  ; Asterisk will use a subdirectory called agi-bin in this
                                            ; directory as the default location for loading AGI scripts
  astspooldir => /var/spool/asterisk        ; The Asterisk spool directory, where voicemail, call
                                            ; recordings, and the call origination spool are stored.
  astrundir => /var/run/asterisk            ; The location where Asterisk will write out it's unix control
                                            ; socket as well as it PID file
  astlogdir => /var/log/asterisk            ; The asterisk log directory

[options]
  verbose = 3                     ; Sets the default verbose setting for the Asterisk logger
  timestamp = yes                 ; Adds timestamps to all output except output from a CLI command
  initcrypto = yes                ; Load keys from the astkeydir at startup
  systemname = jeeves             ; The name of the PBX, yeah ours is called Jeeves, what you going to do about
                                  ; it?
  maxload = 0.8                   ; Sets a maximum load average. If the load average is at or above this
                                  ; threshold, Asterisk will not accept new calls.
  minmemfree = 10                 ; Sets the minimum number of megabytes of free memory required for Asterisk to
                                  ; continue accepting calls.
  cache_record_files = yes        ; When doing recording, stores the file in record_cache_dir until recording is
                                  ; complete.
  runuser = asterisk              ; Run Asterisk under the user specified
  rungroup = asterisk             ; Run Asterisk under the group specified
  defaultlanguage = en            ; Set the default language
  documentation_language = en_US  ; Set the preferred language for the command help

[files]
  astctlpermissions = 0660  ; Sets the permissions for the Asterisk control socket
  astctlowner = root        ; Sets the owner for the Asterisk control socket
  astctlgroup = asterisk    ; Sets the group for the Asterisk control socket
  astctl = asterisk.ctl     ; Sets the filename for the Asterisk control socket

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; --- Call Completion Supplementary Services ---
; Gentlemen

[general]
  cc_max_requests = 20  ; Global limit on the number of CC requests that may be in the Asterisk
                        ; system at any one time.

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; --- ODBC Database CDR storage ---
; Gentlemen

[default]
  connection = gentlemen  ; Name of the connections (references res_odbc.conf)
  table = asterisk        ; Name of the table in the database
  usegmtime = yes         ; Use Greenwich Mean Time in the database

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; --- Asterisk Call Detail Record Engine Configuration ---
; Gentlemen

[general]
  batch = yes
  enable = yes
  safeshutdown = yes
  scheduleronly = no
  size = 100
  time = 300
  unanswered = yes

[csv]
  accountlogs = yes   ; Create separate log file for each account code
  loguniqueid = yes   ; Log uniqueid for the call
  loguserfield = yes  ; Log user field
  usegmtime = yes     ; Log date and time in GMT

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; --- Asterisk Call Detail Records (CDR) - Syslog Backend ---
; Gentlemen

; This line needs to be added to /etc/rsyslog.conf:
; local4.info                   /var/log/asterisk-cdr.log

[cdr]
  facility=local4
  priority=info

  ; High Resolution Time for billsec and duration fields
  template = '${CDR(clid)}','${CDR(src)}','${CDR(dst)}','${CDR(dcontext)}','${CDR(channel)}','${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(start)}','${CDR(answer)}','${CDR(end)}','${CDR(duration,f)}','${CDR(billsec,f)}','${CDR(disposition)}','${CDR(amaflags)}','${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}'

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; --- Asterisk Extension Configuration ---
; Gentlemen

; Global variables to be used in the rest of the dial plan
[globals]
  ; Area information
  LOCALAREACODE=802

  ; User extensions
  U1=SIP/user1
  U2=SIP/user2
  U3=SIP/user3

  ; Device extensions
  PSTNOUT=SIP/SPA3000_PSTNOUT
  PSTNIN=SIP/SPA3000_PSTNIN
  LANDLINE=SIP/SPA3000_LINE

; This context contains the menu for callers coming in over the PSTN line
[pstnmenu]
  exten => s,1,Verbose(3,${CALLERID(all)})  ; Print the Caller ID to the console if it's available
           same => n,GoTo(debug,echo,1)
           same => n,Answer()
           same => n,HangUp()

  exten => i,1,GoTo(s,1)

; This context is where unauthenticated SIP clients from the internet are going
; to end up This will provide a certain measure of spam and robot protection by
; requiring the user to answer a simple math question generated on the fly
[unauthenticated]
  ; If we allow numeric extension calls from the internet:
  ;       exten => _XXX,1,GoTo(internal-public,1,${EXTEN})

  ; If we allow alpha extension calls from the internet (assumes all extensions are lower case)
  ;       exten => _X.,1,Set(SAFE_EXTEN=${FILTER(a-z,${EXTEN})})
  ;               same => n,GoTo(internal-public,1,${SAFE_EXTEN})

  ; Catch any unknown or unallowed extensions and give them a "Network out of order" response
  ; This might trip up some brute force tools, and is better than not passing a reason
  exten => i,1,HangUp(38)

; Our trunk lines should never be making calls into the system, by putting them into this context
; if anyone manages to collect a valid username/password set from one of our trunk they will not be
; able to make calls through our system. This trunk should never have any dialplan in it beyonds 'hang up'
[from-trunk]
  ; Catch any unknown or unallowed extensions and give them a "Network out of order" response
  ; This might trip up some brute force tools, and is better than not passing a reason
  exten => i,1,HangUp(38)

; This is by far the most dangerous context. This one makes outgoing calls from
[callout]
  exten => _NXXNXXXXXX,1,Dial(${PSTNOUT}/1${EXTEN})     ; 10-digit pattern match for NANP
  exten => _9NXXNXXXXXX,1,Dial(${PSTNOUT}/1${EXTEN:1})  ; 10-digit pattern match for NANP (9 first)

  exten => _NXXXXXX,1,Dial(${PSTNOUT}/${LOCALAREACODE}${EXTEN})     ; 7-digit pattern match for NANP
  exten => _9NXXXXXX,1,Dial(${PSTNOUT}/${LOCALAREACODE}${EXTEN:1})  ; 7-digit pattern match for NANP
                                                                    ; (9 first)

  ;exten => _1NXXNXXXXXX,1,Dial(${PSTNOUT}/${EXTEN})     ; Long-distance pattern match for NANP
  ;exten => _91NXXNXXXXXX,1,Dial(${PSTNOUT}/${EXTEN:1})  ; Long-distance pattern match for NANP (9 first)

  ;exten => _011.,1,HangUp(63)   ; Deny the International pattern match for calls made from NANP
                                 ; with 'Service or option unavailable'
  ;exten => _9011.,1,HangUp(63)  ; Deny the International pattern match for calls made from NANP
                                 ; with 'Service or option unavailable' (9 first)

  ; The number they're reaching doesn't match anything we know about, deny it
  exten => i,1,HangUp()

[internal]
  include => callout          ; Allow internal phones to call out
  include => internal-public  ; Include public extensions (user extensions should only be defined once,
                              ; either here or there)

  exten => 600,1,GoTo(debug,echo,1)

; This context holds the private internal extensions that will be available publicly either
; through a PSTN line or through anonymous SIP connections from the internet. These lines
; will be directly dialable from the internal context
[internal-public]
  exten => 301,1,Dial(${U1})
  exten => user1,1,Dial(${U1})

  exten => 302,1,Dial(${U2})
  exten => user2,1,Dial(${U2})

  exten => 303,1,Dial(${U3})
  exten => user3,1,Dial(${U3})

  exten => 304,1,Dial(${LANDLINE})
  exten => landline,1,Dial(${LANDLINE})

; Various extensions that can be used to debug troublesome clients
[debug]
  ; This channel will echo back whatever sounds this server receives from the client
  ; including DTMF tones until they hangup or they press #
  exten => echo,1,Answer()
    same => n,Playback(silence/1)   ; Prevent the beginning of our audio from getting cut off
    same => n,Playback(demo-echotest)
    same => n,Echo()
    same => n,Playback(demo-echodone)
    same => n,Playback(vm-goodbye)
    same => n,Playback(silence/1)   ; Prevent the end of our audio from getting cut off
    same => n,HangUp()

  ; This can be used anywhere in any other dialplan through the GoSub() routine like so:
  ;       exten => example,1,GoSub(debug,log,1(1,[${CHANNEL}] This is the message))
  ; By default all logging is off, to turn it on you need to execute the following commands
  ; at the asterisk console:
  ;       *CLI> core set verbose 0
  ;       *CLI> database put Log all 1
  ;
  ; You can optionally only turn on logging for a single channel by running the following
  ; command in place of the latter command (this example is for the "incoming" channel)
  ;       *CLI> database put Log/channel/incoming 1
  ;
  exten => log,1,GotoIf($[${DB_EXISTS(Log/all)} = 0]?checkchan1)
    same => n,GotoIf($[${ARG1} <= ${DB(Log/all)}]?log)
    same => n(checkchan1),Set(KEY=Log/channel/${CHANNEL})
    same => n,GotoIf($[${DB_EXISTS(${KEY})} = 0]?checkchan2)
    same => n,GotoIf($[${ARG1} <= ${DB(${KEY})}]?log)
    same => n(checkchan2),Set(KEY=Log/channel/${CUT(CHANNEL,-,1)})
    same => n,GotoIf($[${DB_EXISTS(${KEY})} = 0]?return)
    same => n,GotoIf($[${ARG1} <= ${DB(${KEY})}]?log)
    same => n(return),Return()              ; Logging is not turned on return without doing anything
    same => n(log),Verbose(0,${ARG2})       ; Log the message to the console
    same => n,Return()

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; Asterisk Call Features (parking, transfer, etc) Configuration
; Gentlemen

[general]
  ; Nothing has been configured yet

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; --- Logging Configuration ---
; Gentlemen

[general]
  rotatestrategy = rotate
  exec_after_rotate=gzip -9 ${filename}.2

[logfiles]
  console => notice,warning,error,dtmf
  ;full => notice,warning,error,debug,verbose,dtmf,fax
  ;verbose => notice,warning,error,verbose
  messages => notice,warning,error

  syslog.local4 => notice,warning,error

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; --- AMI - The Asterisk Manager Interface Configuration ---
; Gentlemen

[general]
  ;bindaddr = 10.13.37.17
  enabled = no
  ;port = 5038
  webenabled = no

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; --- Module Loader configuration file ---
; Gentlemen

[modules]
  ; I much prefer the whitelist approach of loading modules as I won't get anything that I don't need,
  ; security through simplicity
  autoload=no

  ; Uncomment the following if you wish to use the Speech Recognition API
  ;preload => res_speech.so

  ; Essential modules needed for basic operations
  load => app_dial.so           ; Used to connection channels together (i.e., make phone calls)
  load => app_stack.so          ; Provides GoSub(), GoSubIf(), Return(), StackPop(), LOCAL(),
                                ; and LOCAL_PEEK()
  load => chan_sip.so           ; Session Initiation Protocol channel driver
  load => pbx_config.so         ; This is the traditional, and most popular, dialplan language
                                ; for Asterisk. Without this module, Asterisk cannot read
                                ; extensions.conf
  load => res_rtp_asterisk.so   ; Provides RTP

  ; Security Modules
  load => cdr_adaptive_odbc.so  ; Writes CDRs through ODBC framework to a database
  load => cdr_csv.so            ; Write CDRs to a CSV file
  load => cdr_syslog.so         ; Writes CDRs to syslog
  load => res_security_log.so   ; Enables security logging

  ; Audio Codecs

  ; Recommended Order:
  ;       ulaw, alaw, gsm, g722, g726, speex

  ; Description:          A-law PCM codec used all over the world (except Canada/USA) on the PSTN
  ; Codec:                G.711 (alaw)
  ; Nominal Bandwidth:    64 kbit/s
  ; Algorithmic Latency:  0.125ms
  ; Mean Opinion Score:   4.2
  ; Fax Capable:          Yes
  ; Video Capable:        No
  load => codec_alaw.so

  ; Description:          Global System for Mobile Communications (GSM) codec
  ; Codec:                GSM-FR
  ; Nominal Bandwidth:    13 kbit/s
  ; Algorithmic Latency:
  ; Mean Opinion Score:   3.7
  ; Fax Capable:          No
  ; Video Capable:        No
  load => codec_gsm.so

  ; Description:          Wideband audio codec
  ; Codec:                G.722
  ; Nominal Bandwidth:    64 kbit/s
  ; Algorithmic Latency:
  ; Mean Opinion Score:
  ; Fax Capable:          Maybe
  ; Video Capable:        No
  load => codec_g722.so

  ; Description:          Flavor of ADPCM
  ; Codec:                G.726
  ; Nominal Bandwidth:    16-40 kbit/s
  ; Algorithmic Latency:  0.125ms
  ; Mean Opinion Score:   3.85
  ; Fax Capable:          No
  ; Video Capable:        No
  load => codec_g726.so

  ; Open source speech codec
  ; Codec:                Speex
  ; Nominal Bandwidth:    2.15-22.4 kbit/s
  ; Algorithmic Latency:  30-34ms
  ; Mean Opinion Score:
  ; Fax Capable:          No
  ; Video Capable:        No
  load => codec_speex.so

  ; Description:          Mu-law PCM codec used in Canada/USA on PSTN
  ; Codec:                G.711 (ulaw)
  ; Nominal Bandwidth:    64 kbit/s
  ; Algorithmic Latency:  0.125ms
  ; Mean Opinion Score:   4.2
  ; Fax Capable:          Yes
  ; Video Capable:        No
  load => codec_ulaw.so

  ; Audio Codec Converters
  load => codec_resample.so  ; Re-samples between 8-bit and 16-bit signed linear
  load => codec_a_mu.so      ; A-law to mu-law direct converter

  ; Format Interpreters (used to read audio files from the disk)
  load => format_ogg_vorbis.so  ; Play ogg vorbis files
  load => format_wav.so         ; Play WAV files

  ; Resources & Function Modules
  load => func_cdr.so         ; Used by cdr_syslog module to format it's output
  load => func_strings.so     ; Used for security reasons on incoming calls from the PSTN lines
  load => res_adsi.so         ; Used by voicemail
  load => res_odbc.so         ; Used by the cdr_adaptive_odbc module
  load => res_musiconhold.so  ; Provides music on hold resources

  ; Voicemail
  load => app_voicemail_plain.so  ; Stores voicemails in files

  ; Production Debug Tools
  load => app_echo.so     ; Used in the echo channel to ensure two way audio is working
  load => app_verbose.so  ; Used in the various channels to send messages to the console and logs

  ; Used in various dialplans
  load => app_playback.so  ; Plays back a pre-recorded audio file

  ; For Caller ID information
  load => app_setcallerid.so
  load => func_callerid.so

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; --- Music on Hold Configuration ---
; Gentlemen

[default]
  directory = moh  ; Directory relative to the astdatadir that has the music on hold files
  mode = files     ; Read files from the configured directory
  sort = random    ; Sort the files in random order

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; --- ODBC Configuration File ---
; Gentlemen

; This 'context' name is the name that other files will reference this by
[gentlemen]
  enabled => yes                    ; Not strictly necessary but a quick way to remind myself this can
                                    ; be disabled without deleting or commenting out the whole thing
  dsn => asterisk-gentlemen         ; This value should match an entry in /etc/odbc.ini
  username => databaseuser          ; The username used to connect to the database
  password => databasepass          ; Password for connecting to the database
  pre-connect => yes                ; Open a connection as soon as asterisk starts
  idlecheck => 3600                 ; How often should we check that the database connection is still
                                    ; open? (in seconds)
  share_connections => yes          ; Allow connections to be shared, this reduces overhead but doesn't
                                    ; work on all databases
  limit => 5                        ; What is the maximum number of database connections we can have open
                                    ; at any one time?
  connect_timeout => 5              ; How long should we attempt to connect before considering the
                                    ; connection dead?
  negative_connection_cache => 300  ; When a connection fails, how long should we cache that information
                                    ; before we attempt another connection?

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; --- RTP Configuration ---
; Gentlemen

[general]
  rtpstart=10000
  rtpend=10100

  ;rtpchecksums=no
  ;dtmftimeout=3000
  ;rtcpinterval = 5000    ; Milliseconds between rtcp reports
                          ;(min 500, max 60000, default 5000)

  ; Enable strict RTP protection. This will drop RTP packets that
  ; do not come from the source of the RTP stream. This option is
  ; disabled by default.
  ;strictrtp=yes

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; --- SIP Configuration ---
; Gentlemen

[general]
  ; Features/SIP Defaults
  allowguest = no            ; Disable unauthenticated call
  alwaysauthreject = yes     ; Always respond as if every extension is valid, this makes brute force
                             ; scanning of extensions pointless
  canreinvite = no           ; Force all calls to be relayed through asterisk since SIP clients
                             ; outside the firewall won't be able to directly talk to internal extensions
  context = unauthenticated  ; Default context for incoming calls
  dtmfmode = auto            ; Automatically detect the DTMF tranmission type
  srvlookup = no             ; Disable DNS SRV record lookup on outbound calls
  videosupport = yes         ; May be useful later when setting up video calls

  ; Network options
  bindport = 5060                      ; Explicitely bind to the default SIP port
  externip = x.x.x.x                   ; The current external IP address of The Gentlemens Lounge
  localnet = 10.13.37.0/255.255.255.0  ; Our internal subnet
  tcpenable = yes                      ; We'll allow incoming connections via TCP as well
  udpbindaddr = 10.13.37.xx            ; Listen for UDP requests on the primary interface

  ; Encryption settings
  tlsenable = yes                          ; Turn encryption support on
  tlsbindbindaddr = 10.13.37.xx            ; Listen for encrypted calls on the primary interface
  tlscafile = /var/lib/asterisk/ca.crt     ; Certificate authority's cert
  tlscertfile = /var/lib/asterisk/pbx.pem  ; This server's private key and certificate

  ; Clear the list of codecs
  disallow = all

  ; Set a default order for our codecs, providing all of them will increase compatibility with other
  ; internet clients. Notes on the quality tradeoff can be found in the module.conf file where these
  ; codecs are loaded
  allow = ulaw
  allow = alaw
  allow = gsm
  allow = g722
  allow = g726
  allow = speex
  allow = g729

; Template to restrict SIP users to only the local network
[localonly](!)
  ; Start by denying everyone
  deny = 0.0.0.0/0.0.0.0
  ; Allow connetion that originate from 10.13.37.X to attempt to authenticate against this account
  permit = 10.13.37.0/255.255.255.0

; A template for VoIP extensions that might connect from the outside
[sipclient](!)
  context = internal  ; They should be in our internal context
  host = dynamic      ; The clients will not be static (they could be anywhere)
  type = friend       ; User type
  qualify = yes       ; Talk to them frequently to make sure they're still there (and too hold
                      ; firewall connections open)

; Slight adjustment of the sipclient template to adjust that these extensions will be connecting from the outside
[roadwarrior](!,sipclient)
  nat = yes  ; Yes, they're probably on the other side of our NAT

; This is specifically for my SPA-3000, I've found that even though my device has a static addresses
; I should leave the host type as dynamic as registrations cause errors in the asterisk log, I suppose it
; would be possible to configure the LINE and PSTNIN as peers and turn off registrations in the Sipura's
; configuration, which might be the better route to go. I will have to investigate this in the future but
; this works for now.
[atadevice](!,sipclient,localonly)
  nat = never  ; These devices should never be outside our NAT

; Configuration for internal POTS line on the Linksys SPA-3000
[SPA3000_LINE](atadevice)
  context = internal
  port = 5060
  secret = setyourownpassword

; Configuration for incoming phone calls from the POTS line through the SPA-3000
[SPA3000_PSTNIN](atadevice)
  context = pstnmenu
  secret = setyourownpassword

; Configuration for incoming telephone line on the Linksys SPA-3000
[SPA3000_PSTNOUT](atadevice)
  context = from-trunk
  default-user = from-pbx
  host = 10.13.37.xx
  port = 5061
  secret = setyourownpassword

; SIP Users, these should have especially strong passwords. I use the following to generate
; passwords for them:
;      dd if=/dev/random count=2 bs=8 2>/dev/null | base64 | sed -e 's/=*$//'
; It will create passwords like:
;      9VkLRZz0s9GNFcAica0ONA
;      SOhiPwA0pTupjFx/QCu7BA
;      5Zw+5B8435lrMLKvsSNVpQ
[user1](roadwarrior)
  callerid = "User 1"
  secret = setyourownpassword

[user2](roadwarrior)
  callerid ="User 2"
  secret = setyourownpassword

[user3](roadwarrior)
  callerid = "User 3"
  secret = setyourownpassword

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[asterisk-gentlemen]
Description = Database driver for Asterisk call logs
Trace       = Off
TraceFile   = stderr
Driver      = MySQL
SERVER      = mysqlserver.localhost
PORT        = 3306
DATABASE    = gentlemen

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--
-- Database: `gentlemen`
--

-- --------------------------------------------------------

--
-- Table structure for table `asterisk`
--

DROP TABLE IF EXISTS `asterisk`;
CREATE TABLE `asterisk` (
  `call_id` int(10) unsigned NOT NULL AUTO_INCREMENT,
  `accountcode` varchar(255) DEFAULT NULL COMMENT 'An account ID. This field is user-defined and is empty by default.',
  `src` varchar(255) NOT NULL COMMENT 'The calling party’s caller ID number. It is set automatically and is read-only.',
  `dst` varchar(255) NOT NULL COMMENT 'The destination extension for the call. This field is set automatically and is read-only.',
  `dcontext` varchar(255) NOT NULL COMMENT 'The destination context for the call. This field is set automatically and is read-only.',
  `clid` varchar(255) NOT NULL COMMENT 'The full caller ID, including the name, of the calling party. This field is set automatically and is read-only.',
  `channel` varchar(255) NOT NULL COMMENT 'The calling party’s channel. This field is set automatically and is read-only.',
  `dstchannel` varchar(255) NOT NULL COMMENT 'The called party’s channel. This field is set automatically and is read-only.',
  `lastapp` varchar(255) NOT NULL COMMENT 'The last dialplan application that was executed. This field is set automatically and is read-only.',
  `lastdata` varchar(255) NOT NULL COMMENT 'The arguments passed to the lastapp. This field is set automatically and is read-only.',
  `start` varchar(255) NOT NULL COMMENT 'The start time of the call. This field is set automatically and is read-only.',
  `answer` varchar(255) NOT NULL COMMENT 'The answered time of the call. This field is set automatically and is read-only.',
  `end` varchar(255) NOT NULL COMMENT 'The end time of the call. This field is set automatically and is read-only.',
  `duration` varchar(255) NOT NULL COMMENT 'The number of seconds between the start and end times for the call. This field is set automatically and is read-only.',
  `billsec` varchar(255) NOT NULL COMMENT 'The number of seconds between the answer and end times for the call. This field is set automatically and is read-only.',
  `disposition` varchar(255) NOT NULL COMMENT 'An indication of what happened to the call. This may be NO ANSWER, FAILED, BUSY, ANSWERED, or UNKNOWN.',
  `amaflags` varchar(255) NOT NULL COMMENT 'The Automatic Message Accounting (AMA) flag associated with this call. This may be one of the following: OMIT, BILLING, DOCUMENTATION, or Unknown.',
  `userfield` varchar(255) DEFAULT NULL COMMENT 'A general-purpose user field. This field is empty by default and can be set to a user-defined string.',
  `uniqueid` varchar(255) NOT NULL COMMENT 'The unique ID for the src channel. This field is set automatically and is read-only.',
  PRIMARY KEY (`call_id`)
) ENGINE=MyISAM  DEFAULT CHARSET=latin1 ;