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|) ; 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 ;[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