I am using chan_h323 to connect to h.323 sources. I am using Netmeeting to generate h.323 traffic. When I attempt to call extension 300, I hear no audio coming back from Aterisk, even though it is being sent.
Here are various config files and Atersik CLI output.
h323.conf
; The NuFone Network’s
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 10.50.1.21 ; this SHALL contain a single, valid IP address for this machine
;tos=lowdelay
;
; You may specify a global default AMA flag for iaxtel calls. It must be
; one of ‘default’, ‘omit’, ‘billing’, or ‘documentation’. These flags
; are used in the generation of call detail records.
;
;amaflags = default
;
; You may specify a default account for Call Detail Records in addition
; to specifying on a per-user basis
;
;accountcode=lss0101
;
; You can fine tune codecs here using “allow” and “disallow” clauses
; with specific codecs. Use “all” to represent all formats.
;
disallow=all
;allow=all ; turns on all installed codecs
;disallow=g723.1 ; Hm… Proprietary, don’t use it…
allow=gsm ; Always allow GSM, it’s cool 
allow=ulaw ; 711u
; User-Input Mode (DTMF)
;
; valid entries are: rfc2833, inband
; default is rfc2833
dtmfmode=rfc2833
;
; Default RTP Payload to send RFC2833 DTMF on. This is used to
; interoperate with broken gateways which cannot successfully
; negotiate a RFC2833 payload type in the TerminalCapabilitySet.
;
; You may also specify on either a per-peer or per-user basis below.
;dtmfcodec=101
;
; Set the gatekeeper
; DISCOVER - Find the Gk address using multicast
; DISABLE - Disable the use of a GK
; or - The acutal IP address or hostname of your GK
gatekeeper = DISABLE
;
;
; Tell Asterisk whether or not to accept Gatekeeper
; routed calls or not. Normally this should always
; be set to yes, unless you want to have finer control
; over which users are allowed access to Asterisk.
; Default: YES
;
;AllowGKRouted = yes
;
; Optionally you can determine a user by Source IP versus its H.323 alias.
; Default behavour is to determine user by H.323 alias.
;UserByAlias=no
;
; Default context gets used in siutations where you are using
; the GK routed model or no type=user was found. This gives you
; the ability to either play an invalid message or to simply not
; use user authentication at all.
;
; context=default
;
; H.323 Alias definitions
;
; Type ‘h323’ will register aliases to the endpoint
; and Gatekeeper, if there is one.
;
; Example: if someone calls time@your.asterisk.box.com
; Asterisk will send the call to the extension ‘time’
; in the context default
;
; [default]
; exten => time,1,Answer
; exten => time,2,Playback,current-time
;
; Keyword’s ‘prefix’ and ‘e164’ are only make sense when
; used with a gatekeeper. You can specify either a prefix
; or E.164 this endpoint is responsible for terminating.
;
; Example: The H.323 alias ‘det-gw’ will tell the gatekeeper
; to route any call with the prefix 1248 to this alias. Keyword
; e164 is used when you want to specifiy a full telephone
; number. So a call to the number 18102341212 would be
; routed to the H.323 alias ‘time’.
;
;[time]
;type=h323
;e164=18102341212
;context=default
;
;[det-gw]
;type=h323
;prefix=1248,1313
;context=detroit
;
;
; Inbound H.323 calls from BillyBob would land in the incoming
; context with a maximum of 4 concurrent incoming calls
;
;
; Note: If keyword ‘incominglimit’ are omitted Asterisk will not
; enforce any maximum number of concurrent calls.
;
;[BillyBob]
;type=user
;host=192.168.1.1
;context=incoming
;incominglimit=4
;
;
; Outbound H.323 call to Larry using SlowStart
;
[Larry]
type=peer
host=192.168.2.1
noFastStart=yes
;
[h323ipgwout]
type=peer
host=216.147.158.200
incominglimit=10
[h323ipgwin]
type=user
host=216.147.158.200
context=h323-from-world
incominglimit=10
sip.conf
;
; SIP Configuration example for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/username@domain to call any SIP user on the Internet
; (Don’t forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; 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
;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to ‘osp’
; if asterisk was compiled with OSP support.
realm=obo.globalsat.net ; realm for digest authentication
; defaults to “asterisk”
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
;domain=mydomain.tld ; Set default domain for this host
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; Use “sip show domains” to list local domains
;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
;domain=1.2.3.4 ; Add IP address as local domain
; You can have several “domain” settings
;allowexternalinvites=no ; Disable INVITE and REFER to non-local domains
; Default is yes
;autodomain=yes ; Turn this on to have Asterisk add local host
; name and local IP to domain list.
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility (defaults to “no”)
;tos=184 ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpiry=3600 ; Max length of incoming registration we allow
;defaultexpiry=120 ; Default length of incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to “asterisk”
;videosupport=yes ; Turn on support for SIP video
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ;
;musicclass=default ; Sets the default music on hold class for all SIP calls
; This may also be set for individual users/peers
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we’re not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
; when we’re on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;progressinband=never ; If we should generate in-band ringing always
; use ‘never’ to never use in-band signalling, even in cases
; where some buggy devices might not render it
; Valid values: yes, no, never Default: never
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
; local system will cause loops since SIP is incapable
; of performing a “hairpin” call.
;usereqphone = no ; If yes, “;user=phone” is added to uri that contains
; a valid phone number
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
; info : SIP INFO messages
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
;compactheaders = yes ; send compact sip headers.
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
;notifyringing = yes ; Notify subscriptions on RINGING state
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with ‘401 Unauthorized’
; instead of letting the requester know whether there was
; a matching user or peer for their request
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with
; us. The actual extension is the ‘regexten’ parameter of the registering
; peer or its name if ‘regexten’ is not provided. More than one regexten may
; be supplied if they are separated by ‘&’. Patterns may be used in regexten.
;
;regcontext=sipregistrations
;
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => user[:secret[]]@host[:port][/extension]
;
; If no extension is given, the ‘s’ extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
;
; host is either a host name defined in DNS or the name of a section defined
; below.
;
; Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the ‘s’ extension
;
;
;register => 2345:password@sip_proxy/1234
;
; Register 2345 at sip provider ‘sip_proxy’. Calls from this provider
; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a
; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both directions
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
; 0 = continue forever, hammering the other server until it
; accepts the registration
; Default is 0 tries, continue forever
;callevents=no ; generate manager events when sip ua performs events (e.g. hold)
;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk
; behind a NAT device to communicate with services on the outside.
;externip = 200.201.202.203 ; Address that we’re going to put in outbound SIP messages
; if we’re behind a NAT
; The externip and localnet is used
; when registering and communicating with other proxies
; that we're registered with
;externhost=foo.dyndns.net ; Alternatively you can specify an
; external host, and Asterisk will
; perform DNS queries periodically. Not
; recommended for production
; environments! Use externip instead
;externrefresh=10 ; How often to refresh externhost if
; used
; You may add multiple local networks. A reasonable set of defaults
; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
; The nat= setting is used when Asterisk is on a public IP, communicating with
; devices hidden behind a NAT device (broadband router). If you have one-way
; audio problems, you usually have problems with your NAT configuration or your
; firewall’s support of SIP+RTP ports. You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
;
;nat=no ; Global NAT settings (Affects all peers and users)
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581
; never = Never attempt NAT mode or RFC3581 support
; route = Assume NAT, don’t send rport
; (work around more UNIDEN bugs)
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; as-needed basis? (yes|no)
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
; If set to yes, when a SIP UA registers successfully, the ip address,
; the origination port, the registration period, and the username of
; the UA will be set to database via realtime. If not present, defaults to ‘yes’.
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered? (yes|no|)
; If set to yes, when the registration expires, the friend will vanish from
; the configuration until requested again. If set to an integer,
; friends expire within this number of seconds instead of the
; registration interval.
;ignoreregexpire=yes ; Enabling this setting has two functions:
;
; For non-realtime peers, when their registration expires, the information
; will not be removed from memory or the Asterisk database; if you attempt
; to place a call to the peer, the existing information will be used in spite
; of it having expired
;
; For realtime peers, when the peer is retrieved from realtime storage,
; the registration information will be used regardless of whether
; it has expired or not; if it expires while the realtime peer is still in
; memory (due to caching or other reasons), the information will not be
; removed from realtime storage
; Incoming INVITE and REFER messages can be matched against a list of ‘allowed’
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
; context associated with the user/peer placing the call.
; Domains can be specified using:
; domain=[,]
; Examples:
; domain=myasterisk.dom
; domain=customer.com,customer-context
;
; In addition, all the ‘default’ domains associated with a server should be
; added if incoming request filtering is desired.
; autodomain=yes
;
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no
; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
; non-peers, use your primary domain “identity”
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.
[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can authenticate for outbound calls to other
; realms. We match realm on the proxy challenge and pick an set of
; credentials from this list
; Syntax:
; auth = :@
; auth = #@
; Example:
;auth=mark:topsecret@digium.com
;
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm
;------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; permit permit
; deny deny
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; useclientcode useclientcode
; accountcode accountcode
; setvar setvar
; callerid callerid
; amaflags amaflags
; call-limit call-limit
; restrictcid restrictcid
; subscribecontext subscribecontext
; mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
; sendrpid
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)
;type=peer
;context=from-fwd
;host=fwd.pulver.com
[sipipgwin]
type=peer
context=sip-from-world
host=216.147.131.200
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
;secret=guessit
;username=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;usereqphone=yes ; This provider requires “;user=phone” on URI
;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
[sipipgwout]
type=peer
host=216.147.158.200
disallow=all
allow=ulaw
call-limit=10
;
[sipx]
type=peer
host=216.147.131.234
disallow=all
allow=ulaw
call-limit=10
;
;------------------------------------------------------------------------------
; Definitions of locally connected SIP phones
;
; type = user a device that authenticates to us by “from” field to place calls
; type = peer a device we place calls to or that calls us and we match by host
; type = friend two configurations (peer+user) in one
;
; For local phones, type=friend works most of the time
;
; If you have one-way audio, you propably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
;[grandstream1]
;type=friend
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk
; (1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory)
;mailbox=1234@default ; mailbox 1234 in voicemail context “default”
;disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
;astdb=chan2ext/SIP/grandstream1=1234 ; ensures an astDB entry exists
[301]
type=friend
context=sip-in-local
call-limit=1
canreinvite=no
dtmfmode=rfc2833
host=dynamic
username=301
mailbox=301@sip-in-local
secret=123456
callerid = “Engineering Lab” <301>
[302]
type=friend
context=sip-in-local
call-limit=1
canreinvite=no
dtmfmode=rfc2833
host=dynamic
username=302
mailbox=302@sip-in-local
secret=123456
callerid = “Engineering Lab” <302>
;[xlite1]
; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;type=friend
;regexten=1234 ; When they register, create extension 1234
;callerid=“Jane Smith” <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
;[snom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;secret=blah
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
;language=de ; Use German prompts for this user
;host=dynamic ; This peer register with us
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59 ; IP used until peer registers
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
;vmexten=voicemail ; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to “asterisk”
;restrictcid=yes ; To have the callerid restriced -> sent as ANI
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;[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
; Normally you do NOT need to set this parameter
;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
;secret=blah
;host=dynamic
;insecure=port ; Allow matching of peer by IP address without matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
;qualify=1000 ; Consider it down if it’s 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
;
; Call group and Pickup group should be in the range from 0 to 63
;
;callgroup=1,3-4 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
;defaultip=192.168.0.60 ; IP address to use if peer has not registred
;[cisco1]
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip=192.168.0.4 ; IP address to use until registration
;username=goran ; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
extensions.conf
;
;
[general]
;
autofallthrough=yes
;
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
;
exten => s,1,Dial(SIP/${ARG1},20) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}@sip-in-local) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,n,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}@sip-in-local) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,Goto(default,s,1) ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}@sip-in-local) ; If they press *, send the user into VoicemailMain
exten => a,n,Playback(vm-goodbye)
exten => a,n,Hangup()
;
;
[default]
;
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(digit)=2)
exten => s,n,Set(TIMEOUT(response)=5)
exten => s,n,Background(enter-ext-of-person)
exten => s,n,WaitExten(10)
exten => 300,1,VoicemailMain(@sip-in-local)
exten => 300,n,Playback(vm-goodbye)
exten => 300,n,Hangup()
exten => _3XX,1,Answer()
exten => _3XX,n,Macro(stdexten,${EXTEN})
exten => i,1,playback(pbx-invalid)
exten => i,2,Goto(s,2)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
;
;
[sip-in-local]
ignorepat = 9
exten => 300,1,Answer()
exten => 300,n,VoiceMailMain(@sip-in-local)
exten => 300,n,Hangup()
exten => _3XX,1,Answer()
exten => _3XX,n,Macro(stdexten,${EXTEN})
exten => _4XX,1,Dial(SIP/sipipgwout/${EXTEN},20,r)
exten => _2XX,1,Dial(SIP/sipx/${EXTEN},20,r)
exten => _91800NXXXXXX,1,Dial(SIP/sipipgwout/${EXTEN:1})
exten => _91888NXXXXXX,1,Dial(SIP/sipipgwout/${EXTEN:1})
exten => _91877NXXXXXX,1,Dial(SIP/sipipgwout/${EXTEN:1})
exten => _91866NXXXXXX,1,Dial(SIP/sipipgwout/${EXTEN:1})
exten => _91NXXNXXXXXX,1,Dial(SIP/sipipgwout/${EXTEN:1})
exten => _9011,1,Dial(SIP/sipipgwout/${EXTEN}:1})
exten => i,1,playback(pbx-invalid)
exten => i,2,Hangup()
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
;
[h323-from-world]
exten => 300,1,Answer()
exten => 300,n,VoiceMailMain(@sip-in-local)
exten => 300,n,Hangup()
exten => _3XX,1,Answer()
exten => _3XX,n,Macro(stdexten,${EXTEN})
exten => s,1,Goto(default,s,1)
;
[sip-from-world]
exten => 300,1,Answer()
exten => 300,n,VoiceMailMain(@sip-in-local)
exten => 300,n,Hangup()
exten => _3XX,1,Answer()
exten => _3XX,n,Macro(stdexten,${EXTEN})
exten => s,1,Goto(default,s,1)
CLI
Parsing /etc/asterisk/asterisk.conf
e[0;37;40mParsing /etc/asterisk/extconfig.conf
Asterisk 1.2.8, Copyright © 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘show license’ for details.
=========================================================================
Asterisk Event Logger Started /var/log/asterisk/event_log
Asterisk Dynamic Loader loading preload modules:
e[1;30;40m == e[0;37;40mManager registered action Ping
e[1;30;40m == e[0;37;40mManager registered action Events
e[1;30;40m == e[0;37;40mManager registered action Logoff
e[1;30;40m == e[0;37;40mManager registered action Hangup
e[1;30;40m == e[0;37;40mManager registered action Status
e[1;30;40m == e[0;37;40mManager registered action Setvar
e[1;30;40m == e[0;37;40mManager registered action Getvar
e[1;30;40m == e[0;37;40mManager registered action Redirect
e[1;30;40m == e[0;37;40mManager registered action Originate
e[1;30;40m == e[0;37;40mManager registered action Command
e[1;30;40m == e[0;37;40mManager registered action ExtensionState
e[1;30;40m == e[0;37;40mManager registered action AbsoluteTimeout
e[1;30;40m == e[0;37;40mManager registered action MailboxStatus
e[1;30;40m == e[0;37;40mManager registered action MailboxCount
e[1;30;40m == e[0;37;40mManager registered action ListCommands
Jun 26 14:33:01 e[1;33;40mNOTICEe[0;37;40m[7668]: e[1;37;40mcdr.ce[0;37;40m:e[1;37;40m1191e[0;37;40m e[1;37;40mdo_reloade[0;37;40m: CDR simple logging enabled.
e[1;30;40m == e[0;37;40mRTP Allocating from port range 10000 -> 20000
Asterisk PBX Core Initializing
Registering builtin applications:
e[1;30;40m e[0;37;40m[AbsoluteTimeout]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mAbsoluteTimeoute[0;37;40m’
e[1;30;40m e[0;37;40m[Answer]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mAnswere[0;37;40m’
e[1;30;40m e[0;37;40m[BackGround]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mBackGrounde[0;37;40m’
e[1;30;40m e[0;37;40m[Busy]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mBusye[0;37;40m’
e[1;30;40m e[0;37;40m[Congestion]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mCongestione[0;37;40m’
e[1;30;40m e[0;37;40m[DigitTimeout]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mDigitTimeoute[0;37;40m’
e[1;30;40m e[0;37;40m[Goto]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mGotoe[0;37;40m’
e[1;30;40m e[0;37;40m[GotoIf]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mGotoIfe[0;37;40m’
e[1;30;40m e[0;37;40m[GotoIfTime]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mGotoIfTimee[0;37;40m’
e[1;30;40m e[0;37;40m[ExecIfTime]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mExecIfTimee[0;37;40m’
e[1;30;40m e[0;37;40m[Hangup]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mHangupe[0;37;40m’
e[1;30;40m e[0;37;40m[NoOp]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mNoOpe[0;37;40m’
e[1;30;40m e[0;37;40m[Progress]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mProgresse[0;37;40m’
e[1;30;40m e[0;37;40m[ResetCDR]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mResetCDRe[0;37;40m’
e[1;30;40m e[0;37;40m[ResponseTimeout]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mResponseTimeoute[0;37;40m’
e[1;30;40m e[0;37;40m[Ringing]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mRinginge[0;37;40m’
e[1;30;40m e[0;37;40m[SayNumber]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSayNumbere[0;37;40m’
e[1;30;40m e[0;37;40m[SayDigits]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSayDigitse[0;37;40m’
e[1;30;40m e[0;37;40m[SayAlpha]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSayAlphae[0;37;40m’
e[1;30;40m e[0;37;40m[SayPhonetic]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSayPhonetice[0;37;40m’
e[1;30;40m e[0;37;40m[SetAccount]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetAccounte[0;37;40m’
e[1;30;40m e[0;37;40m[SetAMAFlags]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetAMAFlagse[0;37;40m’
e[1;30;40m e[0;37;40m[SetGlobalVar]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetGlobalVare[0;37;40m’
e[1;30;40m e[0;37;40m[SetLanguage]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetLanguagee[0;37;40m’
e[1;30;40m e[0;37;40m[Set]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSete[0;37;40m’
e[1;30;40m e[0;37;40m[SetVar]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetVare[0;37;40m’
e[1;30;40m e[0;37;40m[ImportVar]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mImportVare[0;37;40m’
e[1;30;40m e[0;37;40m[Wait]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mWaite[0;37;40m’
e[1;30;40m e[0;37;40m[WaitExten]
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mWaitExtene[0;37;40m’
Asterisk Dynamic Loader Starting:
e[1;30;40m e[0;37;40m[e[1;37;40mres_musiconhold.soe[0;37;40m] => (e[33;40mMusic On Hold Resourcee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mMusicOnHolde[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mWaitMusicOnHolde[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetMusicOnHolde[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mStartMusicOnHolde[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application 'e[1;36;40mStopMusicOnHolde[0;37;40m’
Jun 26 14:33:01 e[1;31;40mWARNINGe[0;37;40m[7668]: e[1;37;40mres_musiconhold.ce[0;37;40m:e[1;37;40m856e[0;37;40m e[1;37;40mmoh_registere[0;37;40m: Unable to open pseudo channel for timing… Sound may be choppy.
e[1;30;40m e[0;37;40m[e[1;37;40mres_features.soe[0;37;40m] => (e[33;40mCall Features Resourcee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mParkedCalle[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mParke[0;37;40m’
e[1;30;40m == e[0;37;40mManager registered action ParkedCalls
e[1;30;40m e[0;37;40m[e[1;37;40mres_config_mysql.soe[0;37;40m] => (e[33;40mMySQL RealTime Configuration Drivere[0;37;40m)
Jun 26 14:33:01 e[31;40mERRORe[0;37;40m[7668]: e[1;37;40mres_config_mysql.ce[0;37;40m:e[1;37;40m615e[0;37;40m e[1;37;40mmysql_reconnecte[0;37;40m: MySQL RealTime: Failed to connect database server on . Check debug for more info.
Jun 26 14:33:01 e[1;31;40mWARNINGe[0;37;40m[7668]: e[1;37;40mres_config_mysql.ce[0;37;40m:e[1;37;40m450e[0;37;40m e[1;37;40mload_modulee[0;37;40m: MySQL RealTime: Couldn’t establish connection. Check debug.
Jun 26 14:33:01 e[1;33;40mNOTICEe[0;37;40m[7668]: e[1;37;40mconfig.ce[0;37;40m:e[1;37;40m863e[0;37;40m e[1;37;40mast_config_engine_registere[0;37;40m: Registered Config Engine mysql
MySQL RealTime driver loaded.
e[1;30;40m e[0;37;40m[e[1;37;40mres_agi.soe[0;37;40m] => (e[33;40mAsterisk Gateway Interface (AGI)e[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mDeadAGIe[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mEAGIe[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mAGIe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mres_monitor.soe[0;37;40m] => (e[33;40mCall Monitoring Resourcee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mMonitore[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mStopMonitore[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mChangeMonitore[0;37;40m’
e[1;30;40m == e[0;37;40mManager registered action Monitor
e[1;30;40m == e[0;37;40mManager registered action StopMonitor
e[1;30;40m == e[0;37;40mManager registered action ChangeMonitor
e[1;30;40m e[0;37;40m[e[1;37;40mres_indications.soe[0;37;40m] => (e[33;40mIndications Configuratione[0;37;40m)
e[1;30;40m – e[0;37;40mRegistered indication country ‘at’
e[1;30;40m – e[0;37;40mRegistered indication country ‘au’
e[1;30;40m – e[0;37;40mRegistered indication country ‘br’
e[1;30;40m – e[0;37;40mRegistered indication country ‘be’
e[1;30;40m – e[0;37;40mRegistered indication country ‘ch’
e[1;30;40m – e[0;37;40mRegistered indication country ‘cl’
e[1;30;40m – e[0;37;40mRegistered indication country ‘cn’
e[1;30;40m – e[0;37;40mRegistered indication country ‘cz’
e[1;30;40m – e[0;37;40mRegistered indication country ‘de’
e[1;30;40m – e[0;37;40mRegistered indication country ‘dk’
e[1;30;40m – e[0;37;40mRegistered indication country ‘ee’
e[1;30;40m – e[0;37;40mRegistered indication country ‘es’
e[1;30;40m – e[0;37;40mRegistered indication country ‘fi’
e[1;30;40m – e[0;37;40mRegistered indication country ‘fr’
e[1;30;40m – e[0;37;40mRegistered indication country ‘gr’
e[1;30;40m – e[0;37;40mRegistered indication country ‘hu’
e[1;30;40m – e[0;37;40mRegistered indication country ‘it’
e[1;30;40m – e[0;37;40mRegistered indication country ‘lt’
e[1;30;40m – e[0;37;40mRegistered indication country ‘mx’
e[1;30;40m – e[0;37;40mRegistered indication country ‘nl’
e[1;30;40m – e[0;37;40mRegistered indication country ‘no’
e[1;30;40m – e[0;37;40mRegistered indication country ‘nz’
e[1;30;40m – e[0;37;40mRegistered indication country ‘pl’
e[1;30;40m – e[0;37;40mRegistered indication country ‘pt’
e[1;30;40m – e[0;37;40mRegistered indication country ‘ru’
e[1;30;40m – e[0;37;40mRegistered indication country ‘se’
e[1;30;40m – e[0;37;40mRegistered indication country ‘sg’
e[1;30;40m – e[0;37;40mRegistered indication country ‘uk’
e[1;30;40m – e[0;37;40mRegistered indication country ‘us’
e[1;30;40m – e[0;37;40mRegistered indication country ‘us-o’
e[1;30;40m – e[0;37;40mRegistered indication country ‘tw’
e[1;30;40m – e[0;37;40mRegistered indication country ‘za’
e[1;30;40m – e[0;37;40mSetting default indication country to ‘us’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mPlayTonese[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mStopPlayTonese[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mres_crypto.soe[0;37;40m] => (e[33;40mCryptographic Digital Signaturese[0;37;40m)
e[1;30;40m – e[0;37;40mLoaded PUBLIC key ‘freeworlddialup’
e[1;30;40m – e[0;37;40mLoaded PUBLIC key ‘iaxtel’
e[1;30;40m e[0;37;40m[e[1;37;40mres_adsi.soe[0;37;40m] => (e[33;40mADSI Resourcee[0;37;40m)
e[1;30;40m e[0;37;40m[e[1;37;40mpbx_config.soe[0;37;40m] => (e[33;40mText Extension Configuratione[0;37;40m)
e[1;30;40m e[0;37;40m[e[1;37;40mpbx_spool.soe[0;37;40m] => (e[33;40mOutgoing Spool Supporte[0;37;40m)
e[1;30;40m e[0;37;40m[e[1;37;40mpbx_ael.soe[0;37;40m] => (e[33;40mAsterisk Extension Language Compilere[0;37;40m)
e[1;30;40m e[0;37;40m[e[1;37;40mpbx_dundi.soe[0;37;40m] => (e[33;40mDistributed Universal Number Discovery (DUNDi)e[0;37;40m)
e[1;30;40m == e[0;37;40mUsing TOS bits 0
e[1;30;40m == e[0;37;40mDUNDi Ready and Listening on 0.0.0.0 port 4520
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mDUNDiLookupe[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered custom function DUNDILOOKUP
e[1;30;40m e[0;37;40m[e[1;37;40mpbx_realtime.soe[0;37;40m] => (e[33;40mRealtime Switche[0;37;40m)
e[1;30;40m e[0;37;40m[e[1;37;40mpbx_functions.soe[0;37;40m] => (e[33;40mBuiltin dialplan functionse[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered custom function MD5
e[1;30;40m == e[0;37;40mRegistered custom function CHECK_MD5
e[1;30;40m == e[0;37;40mRegistered custom function MATH
e[1;30;40m == e[0;37;40mRegistered custom function GROUP_COUNT
e[1;30;40m == e[0;37;40mRegistered custom function GROUP_MATCH_COUNT
e[1;30;40m == e[0;37;40mRegistered custom function GROUP
e[1;30;40m == e[0;37;40mRegistered custom function GROUP_LIST
e[1;30;40m == e[0;37;40mRegistered custom function FIELDQTY
e[1;30;40m == e[0;37;40mRegistered custom function REGEX
e[1;30;40m == e[0;37;40mRegistered custom function LEN
e[1;30;40m == e[0;37;40mRegistered custom function STRFTIME
e[1;30;40m == e[0;37;40mRegistered custom function EVAL
e[1;30;40m == e[0;37;40mRegistered custom function CDR
e[1;30;40m == e[0;37;40mRegistered custom function ISNULL
e[1;30;40m == e[0;37;40mRegistered custom function SET
e[1;30;40m == e[0;37;40mRegistered custom function EXISTS
e[1;30;40m == e[0;37;40mRegistered custom function IF
e[1;30;40m == e[0;37;40mRegistered custom function IFTIME
e[1;30;40m == e[0;37;40mRegistered custom function ENV
e[1;30;40m == e[0;37;40mRegistered custom function DB
e[1;30;40m == e[0;37;40mRegistered custom function DB_EXISTS
e[1;30;40m == e[0;37;40mRegistered custom function TIMEOUT
e[1;30;40m == e[0;37;40mRegistered custom function LANGUAGE
e[1;30;40m == e[0;37;40mRegistered custom function MUSICCLASS
e[1;30;40m e[0;37;40m[e[1;37;40mpbx_loopback.soe[0;37;40m] => (e[33;40mLoopback Switche[0;37;40m)
e[1;30;40m e[0;37;40m[e[1;37;40mchan_mgcp.soe[0;37;40m] => (e[33;40mMedia Gateway Control Protocol (MGCP)e[0;37;40m)
e[1;30;40m == e[0;37;40mMGCP Listening on 0.0.0.0:2727
e[1;30;40m == e[0;37;40mUsing TOS bits 0
e[1;30;40m == e[0;37;40mRegistered channel type ‘MGCP’ (Media Gateway Control Protocol (MGCP))
e[1;30;40m e[0;37;40m[e[1;37;40mchan_ooh323.soe[0;37;40m] => (e[33;40mObjective Systems H323 Channele[0;37;40m)
Jun 26 14:33:01 e[1;33;40mNOTICEe[0;37;40m[7668]: e[1;37;40msrc/chan_h323.ce[0;37;40m:e[1;37;40m1752e[0;37;40m e[1;37;40mreload_confige[0;37;40m: Unable to load config ooh323.conf, OOH323 disabled
e[1;30;40m e[0;37;40m[e[1;37;40mchan_features.soe[0;37;40m] => (e[33;40mFeature Proxy Channele[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered channel type ‘Feature’ (Feature Proxy Channel Driver)
e[1;30;40m e[0;37;40m[e[1;37;40mchan_zap.soe[0;37;40m] => (e[33;40mZapata Telephony w/PRIe[0;37;40m)
e[1;30;40m – e[0;37;40mAutomatically generated pseudo channel
e[1;30;40m == e[0;37;40mRegistered channel type ‘Zap’ (Zapata Telephony Driver w/PRI)
e[1;30;40m == e[0;37;40mManager registered action ZapTransfer
e[1;30;40m == e[0;37;40mManager registered action ZapHangup
e[1;30;40m == e[0;37;40mManager registered action ZapDialOffhook
e[1;30;40m == e[0;37;40mManager registered action ZapDNDon
e[1;30;40m == e[0;37;40mManager registered action ZapDNDoff
e[1;30;40m == e[0;37;40mManager registered action ZapShowChannels
e[1;30;40m e[0;37;40m[e[1;37;40mchan_agent.soe[0;37;40m] => (e[33;40mAgent Proxy Channele[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered channel type ‘Agent’ (Call Agent Proxy Channel)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mAgentLogine[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mAgentCallbackLogine[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mAgentMonitorOutgoinge[0;37;40m’
e[1;30;40m == e[0;37;40mManager registered action Agents
e[1;30;40m == e[0;37;40mManager registered action AgentLogoff
e[1;30;40m == e[0;37;40mManager registered action AgentCallbackLogin
e[1;30;40m e[0;37;40m[e[1;37;40mchan_skinny.soe[0;37;40m] => (e[33;40mSkinny Client Control Protocol (Skinny)e[0;37;40m)
e[1;30;40m == e[0;37;40mSkinny listening on 0.0.0.0:2000
e[1;30;40m == e[0;37;40mRegistered channel type ‘Skinny’ (Skinny Client Control Protocol (Skinny))
e[1;30;40m e[0;37;40m[e[1;37;40mchan_h323.soe[0;37;40m] => (e[33;40mThe NuFone Network’s Open H.323 Channel Drivere[0;37;40m)
e[1;30;40m == e[0;37;40mCreating H.323 Endpoint
e[1;30;40m == e[0;37;40mRegistered channel type ‘H323’ (The NuFone Network’s Open H.323 Channel Driver)
e[1;30;40m == e[0;37;40mH.323 listener started
e[1;30;40m e[0;37;40m[e[1;37;40mchan_local.soe[0;37;40m] => (e[33;40mLocal Proxy Channele[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered channel type ‘Local’ (Local Proxy Channel Driver)
e[1;30;40m e[0;37;40m[e[1;37;40mchan_sip.soe[0;37;40m] => (e[33;40mSession Initiation Protocol (SIP)e[0;37;40m)
e[1;30;40m – e[0;37;40mSIP Seeding peer from astdb: ‘301’ at 301@10.50.1.94:5064 for 3600
e[1;30;40m – e[0;37;40mSIP Seeding peer from astdb: ‘302’ at 302@10.50.1.94:5066 for 3600
e[1;30;40m == e[0;37;40mSIP Listening on 0.0.0.0:5060
e[1;30;40m == e[0;37;40mUsing TOS bits 0
e[1;30;40m == e[0;37;40mRegistered channel type ‘SIP’ (Session Initiation Protocol (SIP))
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSIPDtmfModee[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSIPAddHeadere[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSIPGetHeadere[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered custom function SIP_HEADER
e[1;30;40m == e[0;37;40mRegistered custom function SIPPEER
e[1;30;40m == e[0;37;40mRegistered custom function SIPCHANINFO
e[1;30;40m == e[0;37;40mRegistered custom function CHECKSIPDOMAIN
e[1;30;40m == e[0;37;40mManager registered action SIPpeers
e[1;30;40m == e[0;37;40mManager registered action SIPshowpeer
e[1;30;40m e[0;37;40m[e[1;37;40mchan_oss.soe[0;37;40m] => (e[33;40mOSS Console Channel Drivere[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered channel type ‘Console’ (OSS Console Channel Driver)
e[1;30;40m e[0;37;40m[e[1;37;40mchan_iax2.soe[0;37;40m] => (e[33;40mInter Asterisk eXchange (Ver 2)e[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered custom function IAXPEER
Jun 26 14:33:02 e[1;31;40mWARNINGe[0;37;40m[7668]: e[1;37;40mchan_iax2.ce[0;37;40m:e[1;37;40m9581e[0;37;40m e[1;37;40mload_modulee[0;37;40m: Unable to open IAX timing interface: No such file or directory
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mIAX2Provisione[0;37;40m’
e[1;30;40m == e[0;37;40mManager registered action IAXpeers
e[1;30;40m == e[0;37;40mManager registered action IAXnetstats
e[1;30;40m == e[0;37;40mUsing TOS bits 16
e[1;30;40m == e[0;37;40mBinding IAX2 to default address 0.0.0.0:4569
e[1;30;40m – e[0;37;40mdoing lookup for ‘216.207.245.47’
e[1;30;40m == e[0;37;40mRegistered channel type ‘IAX2’ (Inter Asterisk eXchange Driver (Ver 2))
e[1;30;40m == e[0;37;40mIAX Ready and Listening
e[1;30;40m == e[0;37;40mLoaded firmware ‘iaxy.bin’
e[1;30;40m – e[0;37;40mLoaded provisioning template ‘default’
e[1;30;40m e[0;37;40m[e[1;37;40mchan_phone.soe[0;37;40m] => (e[33;40mLinux Telephony API Supporte[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered channel type ‘Phone’ (Standard Linux Telephony API Driver)
e[1;30;40m e[0;37;40m[e[1;37;40mcodec_a_mu.soe[0;37;40m] => (e[33;40mA-law and Mulaw direct Coder/Decodere[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40malawtoulawe[0;37;40m’ from format alaw to ulaw, cost 1
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40mulawtoalawe[0;37;40m’ from format ulaw to alaw, cost 1
e[1;30;40m e[0;37;40m[e[1;37;40mapp_page.soe[0;37;40m] => (e[33;40mPage Multiple Phonese[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mPagee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_privacy.soe[0;37;40m] => (e[33;40mRequire phone number to be entered, if no CallerID sente[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mPrivacyManagere[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_url.soe[0;37;40m] => (e[33;40mSend URL Applicationse[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSendURLe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_dictate.soe[0;37;40m] => (e[33;40mVirtual Dictation Machinee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mDictatee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_disa.soe[0;37;40m] => (e[33;40mDISA (Direct Inward System Access) Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mDISAe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_system.soe[0;37;40m] => (e[33;40mGeneric System() applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mTrySysteme[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSysteme[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_test.soe[0;37;40m] => (e[33;40mInterface Test Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mTestCliente[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mTestServere[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_sendtext.soe[0;37;40m] => (e[33;40mSend Text Applicationse[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSendTexte[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_zapras.soe[0;37;40m] => (e[33;40mZap RAS Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mZapRASe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_meetme.soe[0;37;40m] => (e[33;40mMeetMe conference bridgee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mMeetMeAdmine[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mMeetMeCounte[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mMeetMee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mcodec_ilbc.soe[0;37;40m] => (e[33;40miLBC/PCM16 (signed linear) Codec Translatore[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40milbctoline[0;37;40m’ from format ilbc to slin, cost 2
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40mlintoilbce[0;37;40m’ from format slin to ilbc, cost 15
e[1;30;40m e[0;37;40m[e[1;37;40mapp_realtime.soe[0;37;40m] => (e[33;40mRealtime Data Lookup/Rewritee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mRealTimeUpdatee[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mRealTimee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_playback.soe[0;37;40m] => (e[33;40mSound File Playback Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mPlaybacke[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_random.soe[0;37;40m] => (e[33;40mRandom gotoe[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mRandome[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_getcpeid.soe[0;37;40m] => (e[33;40mGet ADSI CPE IDe[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mGetCPEIDe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_hasnewvoicemail.soe[0;37;40m] => (e[33;40mIndicator for whether a voice mailbox has messages in a given folder.e[)
e[1;30;40m == e[0;37;40mRegistered custom function VMCOUNT
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mHasVoicemaile[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mHasNewVoicemaile[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mformat_au.soe[0;37;40m] => (e[33;40mSun Microsystems AU format (signed linear)e[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format au, extension(s) au
e[1;30;40m e[0;37;40m[e[1;37;40mapp_flash.soe[0;37;40m] => (e[33;40mFlash zap trunk applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mFlashe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_setrdnis.soe[0;37;40m] => (e[33;40mSet RDNIS Numbere[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetRDNISe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_record.soe[0;37;40m] => (e[33;40mTrivial Record Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mRecorde[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mcodec_ulaw.soe[0;37;40m] => (e[33;40mMu-law Coder/Decodere[0;37;40m)
e[1;30;40m – e[0;37;40mcodec_ulaw: using generic PLC
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40mulawtoline[0;37;40m’ from format ulaw to slin, cost 1
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40mlintoulawe[0;37;40m’ from format slin to ulaw, cost 1
e[1;30;40m e[0;37;40m[e[1;37;40mapp_saycountpl.soe[0;37;40m] => (e[33;40mSay polish counting wordse[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSayCountPLe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_softhangup.soe[0;37;40m] => (e[33;40mHangs up the requested channele[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSoftHangupe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mfunc_uri.soe[0;37;40m] => (e[33;40mURI encode/decode functionse[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered custom function URIDECODE
e[1;30;40m == e[0;37;40mRegistered custom function URIENCODE
e[1;30;40m e[0;37;40m[e[1;37;40mfunc_enum.soe[0;37;40m] => (e[33;40mENUM Related Functionse[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered custom function ENUMLOOKUP
e[1;30;40m == e[0;37;40mRegistered custom function TXTCIDNAME
e[1;30;40m e[0;37;40m[e[1;37;40mapp_math.soe[0;37;40m] => (e[33;40mBasic Math Functionse[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mMathe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mcodec_alaw.soe[0;37;40m] => (e[33;40mA-law Coder/Decodere[0;37;40m)
e[1;30;40m – e[0;37;40mcodec_alaw: using generic PLC
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40malawtoline[0;37;40m’ from format alaw to slin, cost 1
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40mlintoalawe[0;37;40m’ from format slin to alaw, cost 1
e[1;30;40m e[0;37;40m[e[1;37;40mformat_mp3.soe[0;37;40m] => (e[33;40mMP3 format [Any rate but 8000hz mono optimal]e[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format mp3, extension(s) mp3
e[1;30;40m e[0;37;40m[e[1;37;40mapp_controlplayback.soe[0;37;40m] => (e[33;40mControl Playback Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mControlPlaybacke[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_transfer.soe[0;37;40m] => (e[33;40mTransfere[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mTransfere[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_voicemail.soe[0;37;40m] => (e[33;40mComedian Mail (Voicemail System)e[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mVoiceMaile[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mVoiceMailMaine[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mMailboxExistse[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mVMAuthenticatee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_directory.soe[0;37;40m] => (e[33;40mExtension Directorye[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mDirectorye[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_while.soe[0;37;40m] => (e[33;40mWhile Loops and Conditional Executione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mWhilee[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mExecIfe[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mEndWhilee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mformat_g723.soe[0;37;40m] => (e[33;40mG.723.1 Simple Timestamp File Formate[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format g723sf, extension(s) g723|g723sf
e[1;30;40m e[0;37;40m[e[1;37;40mformat_wav.soe[0;37;40m] => (e[33;40mMicrosoft WAV format (8000hz Signed Linear)e[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format wav, extension(s) wav
e[1;30;40m e[0;37;40m[e[1;37;40mapp_alarmreceiver.soe[0;37;40m] => (e[33;40mAlarm Receiver for Asteriske[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mAlarmReceivere[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mformat_wav_gsm.soe[0;37;40m] => (e[33;40mMicrosoft WAV format (Proprietary GSM)e[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format wav49, extension(s) WAV|wav49
e[1;30;40m e[0;37;40m[e[1;37;40mformat_g726.soe[0;37;40m] => (e[33;40mRaw G.726 (16/24/32/40kbps) datae[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format g726-40, extension(s) g726-40
e[1;30;40m == e[0;37;40mRegistered file format g726-32, extension(s) g726-32
e[1;30;40m == e[0;37;40mRegistered file format g726-24, extension(s) g726-24
e[1;30;40m == e[0;37;40mRegistered file format g726-16, extension(s) g726-16
e[1;30;40m e[0;37;40m[e[1;37;40mapp_setcidname.soe[0;37;40m] => (e[33;40mSet CallerID Namee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetCIDNamee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_zapbarge.soe[0;37;40m] => (e[33;40mBarge in on Zap channel applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mZapBargee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_userevent.soe[0;37;40m] => (e[33;40mCustom User Event Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mUserEvente[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_sayunixtime.soe[0;37;40m] => (e[33;40mSay timee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSayUnixTimee[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mDateTimee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_parkandannounce.soe[0;37;40m] => (e[33;40mCall Parking and Announce Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mParkAndAnnouncee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mformat_g729.soe[0;37;40m] => (e[33;40mRaw G729 datae[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format g729, extension(s) g729
e[1;30;40m e[0;37;40m[e[1;37;40mapp_nbscat.soe[0;37;40m] => (e[33;40mSilly NBS Stream Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mNBScate[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_chanisavail.soe[0;37;40m] => (e[33;40mCheck channel availabilitye[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mChanIsAvaile[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_txtcidname.soe[0;37;40m] => (e[33;40mTXTCIDNamee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mTXTCIDNamee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mcodec_adpcm.soe[0;37;40m] => (e[33;40mAdaptive Differential PCM Coder/Decodere[0;37;40m)
e[1;30;40m – e[0;37;40mcodec_adpcm: using generic PLC
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40madpcmtoline[0;37;40m’ from format adpcm to slin, cost 1
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40mlintoadpcme[0;37;40m’ from format slin to adpcm, cost 1
e[1;30;40m e[0;37;40m[e[1;37;40mformat_gsm.soe[0;37;40m] => (e[33;40mRaw GSM datae[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format gsm, extension(s) gsm
e[1;30;40m e[0;37;40m[e[1;37;40mapp_waitforring.soe[0;37;40m] => (e[33;40mWaits until first ring after timee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mWaitForRinge[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mformat_vox.soe[0;37;40m] => (e[33;40mDialogic VOX (ADPCM) File Formate[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format vox, extension(s) vox
e[1;30;40m e[0;37;40m[e[1;37;40mapp_queue.soe[0;37;40m] => (e[33;40mTrue Call Queueinge[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mQueuee[0;37;40m’
e[1;30;40m == e[0;37;40mManager registered action Queues
e[1;30;40m == e[0;37;40mManager registered action QueueStatus
e[1;30;40m == e[0;37;40mManager registered action QueueAdd
e[1;30;40m == e[0;37;40mManager registered action QueueRemove
e[1;30;40m == e[0;37;40mManager registered action QueuePause
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mAddQueueMembere[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mRemoveQueueMembere[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mPauseQueueMembere[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mUnpauseQueueMembere[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered custom function QUEUEAGENTCOUNT
e[1;30;40m e[0;37;40m[e[1;37;40mformat_sln.soe[0;37;40m] => (e[33;40mRaw Signed Linear Audio support (SLN)e[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format sln, extension(s) sln|raw
e[1;30;40m e[0;37;40m[e[1;37;40mapp_forkcdr.soe[0;37;40m] => (e[33;40mFork The CDR into 2 separate entities.e[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mForkCDRe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_readfile.soe[0;37;40m] => (e[33;40mStores output of file into a variablee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mReadFilee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mformat_h263.soe[0;37;40m] => (e[33;40mRaw h263 datae[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered file format h263, extension(s) h263
e[1;30;40m e[0;37;40m[e[1;37;40mapp_externalivr.soe[0;37;40m] => (e[33;40mExternal IVR Interface Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mExternalIVRe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_eval.soe[0;37;40m] => (e[33;40mReevaluates stringse[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mEvale[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_cdr.soe[0;37;40m] => (e[33;40mTell Asterisk to not maintain a CDR for the current calle[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mNoCDRe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_lookupcidname.soe[0;37;40m] => (e[33;40mLook up CallerID Name from local databasee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mLookupCIDNamee[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_festival.soe[0;37;40m] => (e[33;40mSimple Festival Interfacee[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mFestivale[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_read.soe[0;37;40m] => (e[33;40mRead Variable Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mReade[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_setcallerid.soe[0;37;40m] => (e[33;40mSet CallerID Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetCallerPrese[0;37;40m’
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSetCallerIDe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_mp3.soe[0;37;40m] => (e[33;40mSilly MP3 Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mMP3Playere[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mapp_senddtmf.soe[0;37;40m] => (e[33;40mSend DTMF digits Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mSendDTMFe[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mfunc_callerid.soe[0;37;40m] => (e[33;40mCaller ID related dialplan functione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered custom function CALLERID
e[1;30;40m e[0;37;40m[e[1;37;40mapp_exec.soe[0;37;40m] => (e[33;40mExecutes applicationse[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mExece[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mcodec_gsm.soe[0;37;40m] => (e[33;40mGSM/PCM16 (signed linear) Codec Translatore[0;37;40m)
e[1;30;40m – e[0;37;40mcodec_gsm: using generic PLC
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40mgsmtoline[0;37;40m’ from format gsm to slin, cost 1
e[1;30;40m == e[0;37;40mRegistered translator ‘e[35;40mlintogsme[0;37;40m’ from format slin to gsm, cost 3
e[1;30;40m e[0;37;40m[e[1;37;40mapp_mixmonitor.soe[0;37;40m] => (e[33;40mMixed Audio Monitoring Applicatione[0;37;40m)
e[1;30;40m == e[0;37;40mRegistered application ‘e[1;36;40mMixMonitore[0;37;40m’
e[1;30;40m e[0;37;40m[e[1;37;40mcdr_addon_mysql.soe[0;37;40m] => (e[33;40mMySQL CDR Backende[0;37;40m)
Jun 26 14:33:02 e[1;31;40mWARNINGe[0;37;40m[7668]: e[1;37;40mcdr_addon_mysql.ce[0;37;40m:e[1;37;40m295e[0;37;40m e[1;37;40mmy_load_modulee[0;37;40m: Unable to load config for mysql CDR’s: cdr_mysql.conf
e[1;30;40m e[0;37;40m[e[1;37;40mcdr_manager.soe[0;37;40m] => (e[33;40mAsterisk Call Manager CDR Backende[0;37;40m)
e[1;30;40m e[0;37;40m[e[1;37;40mapp_setcdruserfield.soe[0;37;40m] => (e[33;40mCDR user field appse[0;37;40m)
e[1;30;4