;; ######### extensions.conf ; extensions.conf - the Asterisk dial plan [general] static=yes writeprotect=no autofallthrough=no clearglobalvars=no [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 [default] ;; nas ;;v exten => 101,1,Dial(SIP/101) exten => 102,1,Dial(SIP/102) exten => 103,1,Dial(SIP/103) exten => 104,1,Dial(SIP/104) ; console exten => 110,1,Dial(SIP/110) exten => 117,1,Dial(SIP/snom) exten => 115,1,Dial(SIP/konn) ; AGI AGI AGI AGI AGI AGI AGI AGI AGI exten => 665,1,Answer() exten => 665,2,AGI(/root/asterisk/agi/test1.php) ;exten => 665,3,Hangup exten => 120,1,Dial(SIP/snom&SIP/konn) exten => 145,1,SayDigits(${EXTEN:1}) ; exten => _6.,1,Dial(mISDN/1/2105156369) ; exten => _6.,1,Dial(mISDN/1/${EXTEN:1}) exten => _7.,1,Dial(mISDN/g:intern/${EXTEN:1}) exten => _8.,1,Dial(mISDN/g:intern/${EXTEN:1}) exten => _9.,1,Dial(SIP/${EXTEN:1}@icall) exten => _.5157753,1,Dial(SIP/103) ;exten => _.X,1,Dial(SIP/110) exten => _.5156369,1,Dial(SIP/110) ; exten => 05157753,1,Dial(SIP/103) ; ##### misdn.conf ; ; 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=2 ; 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 ; 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 ; ; watches the L1s of every port. If one l1 is down it tries to ; get it up. The timeout is given in seconds. with 0 as value it ; does not watch the l1 at all ; ; default value: 0 ; ; this option is only read at loading time of chan_misdn, ; which means you need to unload and load chan_misdn to change the ; value, an asterisk restart should do the trick ; l1watcher_timeout=0 ; 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 ; users sections: ; ; name your sections as you which but not "general" ! ; 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] context=misdn language=gr musicclass=default senddtmf=yes far_alerting=no allowed_bearers=all nationalprefix=0 internationalprefix=00 rxgain=0 txgain=0 te_choose_channel=no pmp_l1_check=no pp_l2_check=no reject_cause=16 need_more_infos=no nttimeout=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 ; ; dialplan means Type Of Number in ISDN Terms (for outgoing calls) ; ; there are different types of the dialplan: ; ; dialplan -> outgoing Number ; localdialplan -> callerid ; cpndialplan -> connected party number ; ; dialplan options: ; ; 0 - unknown ; 1 - International ; 2 - National ; 4 - Subscriber ; ; This setting is used for outgoing calls ; ; default value: 0 ; dialplan=1 localdialplan=1 cpndialplan=0 early_bconnect=yes incoming_early_audio=no always_immediate=yes 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=yes ; 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 presentation=-1 screen=-1 echocancel=no echocancelwhenbridged=no echotraining=no jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=no max_incoming=-1 max_outgoing=-1 ports=1,2,3,4 context=default msns=* ; 2105157754 STUDIO ; 2105157753 FAX ; 2105143212 ; 2105156369 ;[intern] ;ports=1,2,3,4 ;context=intern ;msns=* ;[out] ;ports=1,2,3,4 ;context=from-pstn ;msns=* ;; ######### CLI> OUTPUT dev*CLI> dev*CLI> P[ 3] I IND :SETUP oad:6937351046 dad:5157753 pid:19 state:none P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] --> Bearer: Speech P[ 3] --> Codec: Alaw P[ 0] --> * NEW CHANNEL dad:5157753 oad:6937351046 P[ 3] --> CTON: Unknown P[ 3] --> PRES: Restricted (0) P[ 3] --> SCREEN: Unscreened (0) P[ 3] I SEND:PROCEEDING oad:6937351046 dad:05157753 pid:19 P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] --> TRANSPARENT Mode P[ 3] --> incoming_early_audio off P[ 3] * Starting Ast ctx:default dad:05157753 oad:6937351046 with 's' extension -- Executing Dial("mISDN/3-1", "SIP/110") in new stack -- Called 110 -- SIP/110-081ffe18 is ringing P[ 3] * IND : ringing pid:19 P[ 3] --> * IND : ringing pid:19 P[ 3] I SEND:ALERTING oad:6937351046 dad:05157753 pid:19 P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] --> incoming_early_audio off P[ 3] I IND :RELEASE oad:6937351046 dad:05157753 pid:19 state:ALERTING P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] --> queue_hangup P[ 3] * RELEASING CHANNEL pid:19 ctx:default dad:s oad:6937351046 state: ALERTING P[ 3] I SEND:RELEASE_COMPLETE oad:6937351046 dad:05157753 pid:19 P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad: P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0 P[ 3] $$$ Cleaning up bc with stid :10010300 pid:19 == Spawn extension (default, s, 1) exited non-zero on 'mISDN/3-1' -- Executing Dial("mISDN/3-1", "SIP/110") in new stack -- Called 110 == Spawn extension (default, h, 1) exited non-zero on 'mISDN/3-1' dev*CLI>