; ; chan_misdn sample config ; ; general section: ; ; for debugging and general setup, things that are not bound to port groups ; [general] ; ; Sets the Path to the misdn-init.conf (for nt_ptp mode checking) ; misdn_init=/etc/misdn-init.conf ; 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 ; set debugging file and flags for mISDNuser (NT-Stack) ; ; flags can be or'ed with the following values: ; ; DBGM_NET 0x00000001 ; DBGM_MSG 0x00000002 ; DBGM_FSM 0x00000004 ; DBGM_TEI 0x00000010 ; DBGM_L2 0x00000020 ; DBGM_L3 0x00000040 ; DBGM_L3DATA 0x00000080 ; DBGM_BC 0x00000100 ; DBGM_TONE 0x00000200 ; DBGM_BCDATA 0x00000400 ; DBGM_MAN 0x00001000 ; DBGM_APPL 0x00002000 ; DBGM_ISDN 0x00004000 ; DBGM_SOCK 0x00010000 ; DBGM_CONN 0x00020000 ; DBGM_CDATA 0x00040000 ; DBGM_DDATA 0x00080000 ; DBGM_SOUND 0x00100000 ; DBGM_SDATA 0x00200000 ; DBGM_TOPLEVEL 0x40000000 ; DBGM_ALL 0xffffffff ; ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log ; some pbx systems do cut the L1 for some milliseconds, to avoid ; dropping running calls, we can set this flag to yes and tell ; mISDNuser not to drop the calls on L2_RELEASE ntkeepcalls=no ; the big trace ; ; default value: [not set] ; ;tracefile=/var/log/asterisk/misdn.log ; set to yes if you want mISDN_dsp to bridge the calls in HW ; ; default value: yes ; bridging=no ; stops dialtone after getting first digit on nt Port ; ; default value: yes ; stop_tone_after_first_digit=yes ; whether to append overlapdialed Digits to Extension or not ; ; default value: yes ; append_digits2exten=yes ;;; CRYPTION STUFF ; Whether 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 ;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; SIP channel. Defaults to "no". An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only ; if it is forced and enabled. ; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP ; channel. Defaults to "no". ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is ; resynchronized. Useful to improve the quality of the voice, with ; big jumps in/broken timestamps, usually sent from exotic devices ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP ; channel. Two implementations are currently available - "fixed" ; (with size always equals to jbmaxsize) and "adaptive" (with ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set. ; The option represents the number of milliseconds by which the new ; jitter buffer will pad its size. the default is 40, so without ; modification, the new jitter buffer will set its size to the jitter ; value plus 40 milliseconds. increasing this value may help if your ; network normally has low jitter, but occasionally has spikes. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;----------------------------------------------------------------------------------- ; users sections: ; ; name your sections as you wish but not "general" or "default" ! ; the sections 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=en ; ; This option specifies a default music on hold class to ; use when put on hold if the channel's moh class was not ; explicitly set with Set(CHANNEL(musicclass)=whatever) and ; the peer channel did not suggest a class to use. ; musicclass=default ; ; Either if we should produce DTMF Tones ourselves ; senddtmf=yes ; ; If we should generate Ringing for chan_sip and others ; far_alerting=no ; ; Here you can list which bearer capabilities should be allowed: ; all - allow any bearer capability ; speech - allow speech ; 3_1khz - allow 3.1KHz audio ; digital_unrestricted - allow unrestricted digital ; digital_restricted - allow restricted digital ; video - allow video ; ; Example: ; allowed_bearers=speech,3_1khz ; allowed_bearers=all ; Incoming number prefixes for the indicated Type-Of-Number. These are ; inserted before any number (caller, dialed, connected, redirecting, ; redirection) received from the ISDN link if that number has the ; corresponding Type-Of-Number. ; See the dialplan options. ; ; default values: ; unknownprefix= ; internationalprefix=00 ; nationalprefix=0 ; netspecificprefix= ; subscriberprefix= ; abbreviatedprefix= ; ;unknownprefix= internationalprefix=00 nationalprefix=0 ;netspecificprefix= ;subscriberprefix= ;abbreviatedprefix= ; 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 especially in NL seem to need this set to yes, also in ; switzerland this seems to be important ; ; default value: no ; te_choose_channel=no ; ; Monitors L1 of the port. If L1 is down it tries ; to bring it up. The polling timeout is given in seconds. ; Setting the value to 0 disables monitoring L1 of the port. ; ; default value: 0 ; ; This option is only read at chan_misdn loading time. ; You need to unload and load chan_misdn to change the ; value. An asterisk restart will also do the trick. ; l1watcher_timeout=0 ; ; This option defines, if chan_misdn should check the L1 on a PMP ; before making a group call on it. The L1 may go down for PMP Ports ; so we might need this. ; But be aware! a broken or plugged off cable might be used for a group call ; as well, since chan_misdn has no chance to distinguish if the L1 is down ; because of a lost Link or because the Provider shut it down... ; ; default: no ; pmp_l1_check=no ; ; in PMP this option defines which cause should be sent out to ; the 3. caller. chan_misdn does not support callwaiting on TE ; PMP side. This allows to modify the RELEASE_COMPLETE cause ; at least. ; reject_cause=16 ; ; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING), ; this requests additional Infos, so we can waitfordigits ; without much issues. This works only for PTP Ports ; ; default value: no ; need_more_infos=no ; ; set this to yes if you want to disconnect calls when a timeout occurs ; for example during the overlapdial phase ; nttimeout=no ; Set the method to use for channel selection: ; standard - Use the first free channel starting from the lowest number. ; standard_dec - Use the first free channel starting from the highest 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 ; specify if chan_misdn should collect digits before going into the ; dialplan, you can choose yes=4 Seconds, no, or specify the amount ; of seconds you need; ; overlapdial=yes ; ; dialplan means Type Of Number in ISDN Terms ; There are different types of the dialplan: ; ; dialplan -> for outgoing call's dialed number ; localdialplan -> for outgoing call's callerid ; (if -1 is set use the value from the asterisk channel) ; cpndialplan -> for incoming call's connected party number sent to caller ; (if -1 is set use the value from the asterisk channel) ; ; dialplan options: ; ; 0 - unknown ; 1 - International ; 2 - National ; 3 - Network-Specific ; 4 - Subscriber ; 5 - Abbreviated ; ; default value: 0 ; dialplan=0 localdialplan=0 cpndialplan=0 ; ; turn this to no if you don't mind correct handling of Progress Indicators ; early_bconnect=yes ; ; turn this on if you like to send Tone Indications to a Incoming ; isdn channel on a TE Port. Rarely used, only if the Telco allows ; you to send indications by yourself, normally the Telco sends the ; indications to the remote party. ; ; default: no ; incoming_early_audio=no ; 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 ; ; set this to yes if you want to generate your own dialtone ; with always_immediate=yes, else chan_misdn generates the dialtone ; ; default value: no ; nodialtone=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 ; ; default values: not set = 0 ; range: 0-63 ; ;callgroup=1 ;pickupgroup=1 ; Set the outgoing caller id to the value. ;callerid="name" ; ; these are the exact isdn screening and presentation indicators ; if -1 is given for either value the presentation indicators are used ; from asterisks CALLERPRES function. ; s=0, p=0 -> callerid presented ; s=1, p=1 -> callerid restricted (the remote end does not see it!) ; ; default values s=-1, p=-1 presentation=-1 screen=-1 ; Incoming calls will have a caller ID tag set to this value ; ;incoming_cid_tag = "asterisk" ; With this set, you can automatically append the MSN of a party ; to the cid_tag. Incoming calls have the dialed number appended ; to the tag, and outgoing calls have the caller number appended ; to the tag. An '_' is used to separate the tag from the ; MSN. ; Default is no. ; ;append_msn_to_cid_tag = no ; Select what to do with outgoing COLP information on this port. ; ; 0 - Send out COLP information unaltered. (default) ; 1 - Force COLP to restricted on all outgoing COLP information. ; 2 - Do not send COLP information. outgoing_colp=0 ; Put a display ie in the CONNECT message containing the following ; information if it is available (nt port only): ; ; 0 - Do not put the connected line information in the display ie. ; 1 - Put the available connected line name in the display ie. ; 2 - Put the available connected line number in the display ie. ; 3 - Put the available connected line name and number in the display ie. ; display_connected=0 ; Put a display ie in the SETUP message containing the following ; information if it is available (nt port only): ; ; 0 - Do not put the caller information in the display ie. ; 1 - Put the available caller name in the display ie. ; 2 - Put the available caller number in the display ie. ; 3 - Put the available caller name and number in the display ie. ; display_setup=0 ; This enables echo cancellation with the given number of taps. ; Be aware: Move this setting only to outgoing portgroups! ; A value of zero turns echo cancellation off. ; ; possible values are: 0,32,64,128,256,yes(=128),no(=0) ; ; default value: no ; ;echocancel=no ; ; chan_misdns jitterbuffer, default 4000 ; jitterbuffer=4000 ; ; change this threshold to enable dejitter functionality ; jitterbuffer_upper_threshold=0 ; ; change this to yes, if you want to bridge a mISDN data channel to ; another channel type or to an application. ; hdlc=no ; ; defines the maximum amount of incoming calls per port for ; this group. Calls which exceed the maximum will be marked with ; the channel variable MAX_OVERFLOW. It will contain the amount of ; overflowed calls ; max_incoming=-1 ; ; defines the maximum amount of outgoing calls per port for this group ; exceeding calls will be rejected ; max_outgoing=-1 ; ; Enable/disable the call-completion retention option support (ptp only). ; ; Note: To use the CCBS/CCNR supplementary service feature and other ; supplementary services using FACILITY messages requires a ; modified version of mISDN from: ; http://svn.digium.com/svn/thirdparty/mISDN/trunk ; http://svn.digium.com/svn/thirdparty/mISDNuser/trunk ; cc_request_retention=yes [intern] ; define your ports, e.g. 1,2 (depends on mISDN-driver loading order) ports=1,2 ; context where to go to when incoming Call on one of the above ports context=Intern [internPP] ; ; adding the postfix 'ptp' to a port number is obsolete now, chan_misdn ; parses /etc/misdn-init.conf and sets the ptp mode to the corresponding ; configs. For backwards compatibility you can still set ptp here. ; ports=3 [first_extern] ; again port defs ports=4 ; again a context for incoming 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 separated list or simply an '*' for ; any msn. msns=* ; here an example with given msns [second_extern] ports=5 context=Extern2 callerid="Asterisk" <1234> msns=102,144,101,104