EXTENSIONS.CONF
[intern]
#include "clients.conf"
;exten => 8001,1,Dial(Sip/home_del)
;exten => 8002,1,Dial(Sip/home_del)
;exten => 8003,1,Dial(Sip/Anette)
;exten => 8004,1,Dial(Sip/home_del)
;exten => 8005,1,Dial(Sip/home_del)
;exten => 8006,1,Dial(Sip/home_del)
;exten => 8007,1,Dial(Sip/home_del)
exten => 8008,1,Answer()
exten => 8008,2,Background(enter-ext-of-person)
exten => 8008,3,Background(silence/10)
exten => 8008,4,Background(enter-ext-of-person)
exten => 8008,5,Background(silence/10)
exten => 8008,6,Background(enter-ext-of-person)
exten => 8008,7,Background(silence/10)
exten => 1,1,Goto(intern,35,1)
exten => 2,1,Goto(intern,40,1)
exten => 3,1,Goto(intern,60,1)
exten => 4,1,Goto(intern,31,1)
exten => 5,1,VoiceMail(35@default)
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(intern,8008,1)
exten => t,1,Playback(vm-goodbye)
exten => t,2,HangUp()
;exten => 8009,1,Dial(Sip/home_del)
;exten => 8010,1,Dial(Sip/home_del)
;exten => 8011,1,Dial(Sip/home_del)
;exten => 8012,1,Dial(Sip/home_del)
;exten => 8013,1,Dial(Sip/home_del)
;exten => 8014,1,Dial(Sip/home_del)
exten => _90[123458]XXXXXXXX,1,Dial(mISDN/3/${EXTEN:1})
exten => _906XXXXXXXX,1,ChanIsAvail(Zap/26)
exten => _906XXXXXXXX,2,NoOp(${AVAILORIGCHAN})
exten => _906XXXXXXXX,3,NoOp(Disponibilite de Zap/26 = "${AVAILORIGCHAN}")
exten => _906XXXXXXXX,4,Gotoif($["${AVAILORIGCHAN}"="Zap/26"]?5:7)
exten => _906XXXXXXXX,5,Macro(portables3,${EXTEN:2})
exten => _906XXXXXXXX,6,Goto(8)
exten => _906XXXXXXXX,7,Dial(mISDN/g:intern/${EXTEN:1})
exten => _906XXXXXXXX,8,NoOp
;
; 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=mydomain.tld ; 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 = phonesystems2 ; 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,
limitonpeers=yes ; 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[:authuser]]@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 = 213.44.169.43 ;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=yes ; 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|<seconds>)
; 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=<domain>[,<context>]
; 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 = <user>:<secret>@<realm>
; auth = <user>#<md5secret>@<realm>
; 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
;[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
;------------------------------------------------------------------------------
; 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
;[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
register => 33811034119:@213.11.62.39/33811034119
register => 33811035296:@213.11.62.39/33811035296
register => 33811034458:@213.11.62.39/33811034458
;register => test:test@sip.sipdiscount.com/test
;[test]
;type=peer
;username=test
;secret=test
;host=sip.sipdiscount.com
;fromuser=test
;disallow=all
;allow=ulaw
;allow=alaw
;context=office
;dtmfmode=rfc2833
;qualify=yes
;nat=yes
;fromdomain=sipdiscount.com
;callerid=test
[phonesystems3]
type=peer
username=33811034458
secret=
host=213.11.62.39
fromuser=33811034458
disallow=all
allow=ulaw
allow=alaw
context=phonesystems3
dtmfmode=rfc2833
qualify=yes
nat=yes
fromdomain=213.11.62.39
callerid=33811034458
[phonesystems2]
type=peer
username=33811035296
secret=
host=213.11.62.39
fromuser=33811035296
disallow=all
allow=ulaw
allow=alaw
context=phonesystems2
dtmfmode=rfc2833
qualify=yes
nat=yes
fromdomain=213.11.62.39
callerid=33811035296
[phonesystems]
type=peer
username=33811034119
secret=
host=213.11.62.39
fromuser=33811034119
disallow=all
allow=ulaw
allow=alaw
context=phonesystems
dtmfmode=rfc2833
qualify=yes
nat=yes
fromdomain=213.11.62.39
[Anette]
type=friend
secret=
username=243
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
context=Anette
qualify=100
callerid=Anette <243>
call-limit=1
[Thierry]
type=friend
secret=
username=239
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
context=Thierry
qualify=100
callerid=Atelier <239>
call-limit=1
[Tania]
type=friend
secret=
username=217
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
context=phonesystems3
qualify=150
callerid=Tania <217>
[Accueil]
type=friend
secret=
username=35
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
context=phonesystems2
qualify=150
callerid= Accueil <35>
call-limit=5
mailbox=30@default
callgroup=1
pickupgroup=1
[40]
type=friend
secret=
username=40
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
context=phonesystems3
qualify=150
callerid= "CatheB" <40>
call-limit=5
mailbox=40@default
callgroup=1
pickupgroup=1
[31]
type=friend
secret=
username=31
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
context=phonesystems3
qualify=150
callerid="Dietrich" <40>
call-limit=5
mailbox=31@default
callgroup=1
pickupgroup=1
[local]
type=friend
secret=
username=301
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
context=intern
qualify=150
callerid=Admin <301>
call-limit=2
mailbox=888@default
callgroup=1
pickupgroup=1
[home_del]
type=friend
secret=
username=242
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=no
context=phonesystems3
qualify=500
callerid=Domicile Del <242>
call-limit=1
;
; chan_misdn sample config
;
; general section:
;
; for debugging and general setup, things that are not bound to port groups
;
[general]
; set debugging flag:
; 0 - No Debug
; 1 - mISDN Messages and * - Messages, and * - State changes
; 2 - Messages + Message specific Informations (e.g. bearer capability)
; 3 - very Verbose, the above + lots of Driver specific infos
; 4 - even more Verbose than 3
;
; default value: 0
;
debug=0
; the big trace
;
; default value: [not set]
;
;tracefile=/var/log/misdn.trace
; single call trace files
; set to true if you want to have them
; they depend on debug level
;
; default values: trace_calls : false
; trace_dir : /var/log/
;
trace_calls=false
trace_dir=/var/log/
; set to yes if you want mISDN_dsp to bridge the calls in HW
;
; default value: yes
;
bridging=yes
; stops dialtone after getting first digit on nt Port
;
; default value: yes
;
stop_tone_after_first_digit=yes
; wether to append overlapdialed Digits to Extension or not
;
; default value: yes
;
append_digits2exten=yes
; set this to yes if you have jollys mISDN which sends correct L1 Infos
;
; default value: yes
;
l1_info_ok=yes
; set this to yes if you want to clear the l3 in case the l2 deactivates
; some environments have a flickering l2 which causes this option to
; damage active calls .. highly experimental
;
; default value: no
;
clear_l3=no
; set the method to use for channel selection:
; standard - always choose the first free channel with the lowest number
; round_robin - use the round robin algorithm to select a channel. use this
; if you want to balance your load.
;
; default value: standard
;
method=standard
;;; CRYPTION STUFF
; Wether to look for dynamic crypting attempt
;
; default value: no
;
dynamic_crypt=no
; crypt_prefix, what is used for crypting Protocol
;
; default value: [not set]
;
crypt_prefix=**
; Keys for cryption, you reference them in the dialplan
; later also in dynamic encr.
;
; default value: [not set]
;
crypt_keys=test,muh
; users sections:
;
; name your sections as you which but not "general" !
; the secions are Groups, you can dial out in extensions.conf
; with Dial(mISDN/g:extern/101) where extern is a section name,
; chan_misdn tries every port in this section to find a
; new free channel
;
; The default section is not a group section, it just contains config elements
; which are inherited by group sections.
;
[default]
; define your default context here
;
; default value: default
;
context=misdn
; language
;
; default value: en
;
language=fr
; Prefixes for national and international, those are put before the
; oad if an according dialplan is set by the other end.
;
; default values: nationalprefix : 0
; internationalprefix : 00
;
nationalprefix=0
internationalprefix=00
; set rx/tx gains between -8 and 8 to change the RX/TX Gain
;
; default values: rxgain: 0
; txgain: 0
;
rxgain=0
txgain=0
; some telcos espacially in NL seem to need this set to yes, also in
; switzerland this seems to be important
;
; default value: no
;
te_choose_channel=no
; dialplan options:
;
; 0 - unknown
; 1 - National
; 2 - International
; 4 - Subscriber
;
; This setting is used for outgoing calls
;
; default value: 0
;
dialplan=0
; This is only for asterisk head and will result in only considering
; misdn.confs and misdn_set_opts callingpresentation informations if set to no.
; Otherwise asterisks callingpresentation overwrites misdn.confs settings.
;
; default value: yes
;
use_callingpres=yes
; uncomment the following to get into s extension at extension conf
; there you can use DigitTimeout if you can't or don't want to use
; isdn overlap dial.
; note: This will jump into the s exten for every exten!
;
; default value: no
;
;always_immediate=no
; uncomment the following if you want callers which called exactly the
; base number (so no extension is set) jump to the s extension.
; if the user dials something more it jumps to the correct extension
; instead
;
; default value: no
;
;immediate=no
; uncomment the following to have hold and retrieve support
;
; default value: no
;
;hold_allowed=yes
; Pickup and Callgroup
;
; deafult values: not set = 0
;
;callgroup=1
;pickupgroup=1
; Allows/Screens Callerid
;
; possible values: allowed,not_screened
;
; be aware, if you set to allowed you need to set a correct
; callerid in the dialplan or set it here in the misdn.conf
; Some Telcos don't care about wrong callerids, others do !
;
; default value: allowed
;
;presentation=not_screened
; this enables echocancellation, with the given number of taps
; be aware, move this setting only to outgoing portgroups!
; A value of zero turns echocancellation off.
;
; possible values are: 0,32,64,128,256,yes(=128),no(=0)
;
; default value: no
;
;echocancel=no
; this disables echocancellation when the call is bridged between
; mISDN channels
;
; default value: no
;
echocancelwhenbridged=no
; Set this to no to disable echotraining
;
; default value: yes
;
echotraining=yes
[intern]
; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
ports=1,2,3,4
; context where to go to when incoming Call on one of the above ports
context=intern
callerid=asreceived
echocancelwhenbridged=yes
msns=*
;[internPP]
; if you want to have pp Protocol on one nt Port, you need
; to add a ptp directly after the portnumber, you can still add
; more ports and multiple ptp adds in your config.
;ports=3ptp
;[first_extern]
; again port defs
;ports=4
; again a context for incomming calls
;context=Extern1
; msns for te ports, listen on those numbers on the above ports, and
; indicate the incoming calls to asterisk
; here you can give a comma seperated list or simply an '*' for
; any msn.
;msns=*
; here an example with given msns
;[second_extern]
;ports=5
;context=Extern2
;callerid=15
;msns=102,144,101,104