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538 lines
14 KiB
Plaintext
538 lines
14 KiB
Plaintext
;
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; chan_misdn sample config
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;
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; general section:
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;
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; for debugging and general setup, things that are not bound to port groups
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;
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[general]
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;
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; Sets the Path to the misdn-init.conf (for nt_ptp mode checking)
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;
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misdn_init=/etc/misdn-init.conf
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; set debugging flag:
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; 0 - No Debug
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; 1 - mISDN Messages and * - Messages, and * - State changes
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; 2 - Messages + Message specific Informations (e.g. bearer capability)
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; 3 - very Verbose, the above + lots of Driver specific infos
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; 4 - even more Verbose than 3
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;
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; default value: 0
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;
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debug=0
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; set debugging file and flags for mISDNuser (NT-Stack)
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;
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; flags can be or'ed with the following values:
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;
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; DBGM_NET 0x00000001
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; DBGM_MSG 0x00000002
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; DBGM_FSM 0x00000004
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; DBGM_TEI 0x00000010
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; DBGM_L2 0x00000020
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; DBGM_L3 0x00000040
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; DBGM_L3DATA 0x00000080
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; DBGM_BC 0x00000100
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; DBGM_TONE 0x00000200
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; DBGM_BCDATA 0x00000400
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; DBGM_MAN 0x00001000
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; DBGM_APPL 0x00002000
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; DBGM_ISDN 0x00004000
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; DBGM_SOCK 0x00010000
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; DBGM_CONN 0x00020000
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; DBGM_CDATA 0x00040000
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; DBGM_DDATA 0x00080000
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; DBGM_SOUND 0x00100000
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; DBGM_SDATA 0x00200000
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; DBGM_TOPLEVEL 0x40000000
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; DBGM_ALL 0xffffffff
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;
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ntdebugflags=0
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ntdebugfile=/var/log/misdn-nt.log
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; some pbx systems do cut the L1 for some milliseconds, to avoid
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; dropping running calls, we can set this flag to yes and tell
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; mISDNuser not to drop the calls on L2_RELEASE
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ntkeepcalls=no
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; the big trace
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;
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; default value: [not set]
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;
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;tracefile=/var/log/asterisk/misdn.log
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; set to yes if you want mISDN_dsp to bridge the calls in HW
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;
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; default value: yes
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;
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bridging=no
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; stops dialtone after getting first digit on nt Port
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;
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; default value: yes
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;
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stop_tone_after_first_digit=yes
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; whether to append overlapdialed Digits to Extension or not
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;
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; default value: yes
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;
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append_digits2exten=yes
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;;; CRYPTION STUFF
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; Whether to look for dynamic crypting attempt
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;
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; default value: no
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;
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dynamic_crypt=no
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; crypt_prefix, what is used for crypting Protocol
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;
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; default value: [not set]
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;
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crypt_prefix=**
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; Keys for cryption, you reference them in the dialplan
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; later also in dynamic encr.
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;
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; default value: [not set]
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;
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crypt_keys=test,muh
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; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
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; SIP channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The SIP channel can accept jitter,
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; thus a jitterbuffer on the receive SIP side will be used only
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; if it is forced and enabled.
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; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
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; channel. Defaults to "no".
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; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmaxsize) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
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; The option represents the number of milliseconds by which the new
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; jitter buffer will pad its size. the default is 40, so without
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; modification, the new jitter buffer will set its size to the jitter
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; value plus 40 milliseconds. increasing this value may help if your
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; network normally has low jitter, but occasionally has spikes.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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; ----------------------------------------------------------------------------------
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; users sections:
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;
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; name your sections as you wish but not "general" or "default" !
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; the sections are Groups, you can dial out in extensions.conf
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; with Dial(mISDN/g:extern/101) where extern is a section name,
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; chan_misdn tries every port in this section to find a
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; new free channel
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;
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; The default section is not a group section, it just contains config elements
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; which are inherited by group sections.
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;
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[default]
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; define your default context here
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;
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; default value: default
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;
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context=misdn
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; language
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;
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; default value: en
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;
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language=en
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;
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; This option specifies a default music on hold class to
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; use when put on hold if the channel's moh class was not
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; explicitly set with Set(CHANNEL(musicclass)=whatever) and
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; the peer channel did not suggest a class to use.
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;
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musicclass=default
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;
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; Either if we should produce DTMF Tones ourselves
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;
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senddtmf=yes
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;
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; If we should generate Ringing for chan_sip and others
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;
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far_alerting=no
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;
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; Here you can list which bearer capabilities should be allowed:
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; all - allow any bearer capability
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; speech - allow speech
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; 3_1khz - allow 3.1KHz audio
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; digital_unrestricted - allow unrestricted digital
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; digital_restricted - allow restricted digital
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; video - allow video
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;
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; Example:
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; allowed_bearers=speech,3_1khz
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;
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allowed_bearers=all
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; Incoming number prefixes for the indicated Type-Of-Number. These are
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; inserted before any number (caller, dialed, connected, redirecting,
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; redirection) received from the ISDN link if that number has the
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; corresponding Type-Of-Number.
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; See the dialplan options.
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;
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; default values:
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; unknownprefix=
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; internationalprefix=00
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; nationalprefix=0
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; netspecificprefix=
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; subscriberprefix=
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; abbreviatedprefix=
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;
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;unknownprefix=
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internationalprefix=00
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nationalprefix=0
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;netspecificprefix=
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;subscriberprefix=
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;abbreviatedprefix=
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; set rx/tx gains between -8 and 8 to change the RX/TX Gain
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;
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; default values: rxgain: 0
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; txgain: 0
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;
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rxgain=0
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txgain=0
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; some telcos especially in NL seem to need this set to yes, also in
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; switzerland this seems to be important
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;
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; default value: no
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;
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te_choose_channel=no
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;
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; Monitors L1 of the port. If L1 is down it tries
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; to bring it up. The polling timeout is given in seconds.
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; Setting the value to 0 disables monitoring L1 of the port.
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;
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; default value: 0
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;
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; This option is only read at chan_misdn loading time.
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; You need to unload and load chan_misdn to change the
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; value. An asterisk restart will also do the trick.
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;
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l1watcher_timeout=0
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;
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; This option defines, if chan_misdn should check the L1 on a PMP
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; before making a group call on it. The L1 may go down for PMP Ports
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; so we might need this.
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; But be aware! a broken or plugged off cable might be used for a group call
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; as well, since chan_misdn has no chance to distinguish if the L1 is down
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; because of a lost Link or because the Provider shut it down...
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;
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; default: no
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;
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pmp_l1_check=no
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;
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; in PMP this option defines which cause should be sent out to
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; the 3. caller. chan_misdn does not support callwaiting on TE
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; PMP side. This allows to modify the RELEASE_COMPLETE cause
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; at least.
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;
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reject_cause=16
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;
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; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING),
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; this requests additional Infos, so we can waitfordigits
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; without much issues. This works only for PTP Ports
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;
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; default value: no
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;
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need_more_infos=no
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;
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; set this to yes if you want to disconnect calls when a timeout occurs
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; for example during the overlapdial phase
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;
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nttimeout=no
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; Set the method to use for channel selection:
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; standard - Use the first free channel starting from the lowest number.
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; standard_dec - Use the first free channel starting from the highest number.
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; round_robin - Use the round robin algorithm to select a channel. Use this
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; if you want to balance your load.
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;
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; default value: standard
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;
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method=standard
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; specify if chan_misdn should collect digits before going into the
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; dialplan, you can choose yes=4 Seconds, no, or specify the amount
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; of seconds you need;
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;
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overlapdial=yes
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;
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; dialplan means Type Of Number in ISDN Terms
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; There are different types of the dialplan:
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;
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; dialplan -> for outgoing call's dialed number
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; localdialplan -> for outgoing call's callerid
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; (if -1 is set use the value from the asterisk channel)
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; cpndialplan -> for incoming call's connected party number sent to caller
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; (if -1 is set use the value from the asterisk channel)
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;
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; dialplan options:
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;
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; 0 - unknown
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; 1 - International
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; 2 - National
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; 3 - Network-Specific
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; 4 - Subscriber
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; 5 - Abbreviated
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;
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; default value: 0
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;
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dialplan=0
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localdialplan=0
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cpndialplan=0
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;
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; turn this to no if you don't mind correct handling of Progress Indicators
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;
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early_bconnect=yes
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;
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; turn this on if you like to send Tone Indications to a Incoming
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; isdn channel on a TE Port. Rarely used, only if the Telco allows
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; you to send indications by yourself, normally the Telco sends the
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; indications to the remote party.
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;
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; default: no
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;
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incoming_early_audio=no
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; uncomment the following to get into s extension at extension conf
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; there you can use DigitTimeout if you can't or don't want to use
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; isdn overlap dial.
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; note: This will jump into the s exten for every exten!
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;
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; default value: no
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;
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;always_immediate=no
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;
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; set this to yes if you want to generate your own dialtone
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; with always_immediate=yes, else chan_misdn generates the dialtone
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;
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; default value: no
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;
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nodialtone=no
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; uncomment the following if you want callers which called exactly the
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; base number (so no extension is set) jump to the s extension.
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; if the user dials something more it jumps to the correct extension
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; instead
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;
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; default value: no
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;
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;immediate=no
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; uncomment the following to have hold and retrieve support
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;
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; default value: no
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;
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;hold_allowed=yes
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; Pickup and Callgroup
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;
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; default values: not set = 0
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; range: 0-63
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;
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;callgroup=1
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;pickupgroup=1
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; Named pickup groups and named call groups
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;
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; give a name to groups and configure any number of groups
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;
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;namedcallgroup=engineering,sales,netgroup,protgroup
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;namedpickupgroup=sales
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; Set the outgoing caller id to the value.
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;callerid="name" <number>
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;
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; these are the exact isdn screening and presentation indicators
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; if -1 is given for either value the presentation indicators are used
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; from asterisks CALLERPRES function.
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; s=0, p=0 -> callerid presented
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; s=1, p=1 -> callerid restricted (the remote end does not see it!)
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;
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; default values s=-1, p=-1
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presentation=-1
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screen=-1
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; Incoming calls will have a caller ID tag set to this value
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;
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;incoming_cid_tag = "asterisk"
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; With this set, you can automatically append the MSN of a party
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; to the cid_tag. Incoming calls have the dialed number appended
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; to the tag, and outgoing calls have the caller number appended
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; to the tag. An '_' is used to separate the tag from the
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; MSN.
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; Default is no.
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;
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;append_msn_to_cid_tag = no
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; Select what to do with outgoing COLP information on this port.
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;
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; 0 - Send out COLP information unaltered. (default)
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; 1 - Force COLP to restricted on all outgoing COLP information.
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; 2 - Do not send COLP information.
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outgoing_colp=0
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; Put a display ie in the CONNECT message containing the following
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; information if it is available (nt port only):
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;
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; 0 - Do not put the connected line information in the display ie.
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; 1 - Put the available connected line name in the display ie.
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; 2 - Put the available connected line number in the display ie.
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; 3 - Put the available connected line name and number in the display ie.
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;
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display_connected=0
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; Put a display ie in the SETUP message containing the following
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; information if it is available (nt port only):
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;
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; 0 - Do not put the caller information in the display ie.
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; 1 - Put the available caller name in the display ie.
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; 2 - Put the available caller number in the display ie.
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; 3 - Put the available caller name and number in the display ie.
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;
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display_setup=0
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; This enables echo cancellation with the given number of taps.
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; Be aware: Move this setting only to outgoing portgroups!
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; A value of zero turns echo cancellation off.
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;
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; possible values are: 0,32,64,128,256,yes(=128),no(=0)
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;
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; default value: no
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;
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;echocancel=no
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;
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; chan_misdns jitterbuffer, default 4000
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;
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jitterbuffer=4000
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;
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; change this threshold to enable dejitter functionality
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;
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jitterbuffer_upper_threshold=0
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;
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; change this to yes, if you want to bridge a mISDN data channel to
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; another channel type or to an application.
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;
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hdlc=no
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;
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; defines the maximum amount of incoming calls per port for
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; this group. Calls which exceed the maximum will be marked with
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; the channel variable MAX_OVERFLOW. It will contain the amount of
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; overflowed calls
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;
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max_incoming=-1
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;
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; defines the maximum amount of outgoing calls per port for this group
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; exceeding calls will be rejected
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;
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max_outgoing=-1
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;
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; Enable/disable the call-completion retention option support (ptp only).
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;
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; Note: To use the CCBS/CCNR supplementary service feature and other
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; supplementary services using FACILITY messages requires a
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; modified version of mISDN from:
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; http://svn.digium.com/svn/thirdparty/mISDN/trunk
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; http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
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;
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cc_request_retention=yes
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[intern]
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; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
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ports=1,2
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; context where to go to when incoming Call on one of the above ports
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context=Intern
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[internPP]
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;
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; adding the postfix 'ptp' to a port number is obsolete now, chan_misdn
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; parses /etc/misdn-init.conf and sets the ptp mode to the corresponding
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; configs. For backwards compatibility you can still set ptp here.
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;
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ports=3
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[first_extern]
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; again port defs
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ports=4
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; again a context for incoming calls
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context=Extern1
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; msns for te ports, listen on those numbers on the above ports, and
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; indicate the incoming calls to asterisk
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; here you can give a comma separated list or simply an '*' for
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; any msn.
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msns=*
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; here an example with given msns
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[second_extern]
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ports=5
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context=Extern2
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callerid="Asterisk" <1234>
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msns=102,144,101,104
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