5c4578f4ad
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
538 lines
14 KiB
Text
538 lines
14 KiB
Text
;
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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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