updated extensions.conf;; ######### extensions.conf; extensions.conf - the Asterisk dial plan[general]static=yeswriteprotect=noautofallthrough=noclearglobalvars=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); consoleexten => 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,Hangupexten => 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/${EXTEN:1}@icall);exten => _.5157753,1,Dial(SIP/103);make sure 'set verbose 10' is executed and the below should place these values in the console for youexten => _.X,1,NoOp(${calleridnum})exten => _.X,2,NoOp(${exten});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=0ntdebugfile=/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=misdnlanguage=grmusicclass=defaultsenddtmf=yesfar_alerting=noallowed_bearers=allnationalprefix=0internationalprefix=00rxgain=0txgain=0te_choose_channel=nopmp_l1_check=nopp_l2_check=noreject_cause=16need_more_infos=nonttimeout=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=1localdialplan=1cpndialplan=0early_bconnect=yesincoming_early_audio=noalways_immediate=yesnodialtone=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=1presentation=-1screen=-1echocancel=noechocancelwhenbridged=noechotraining=nojitterbuffer=4000jitterbuffer_upper_threshold=0hdlc=nomax_incoming=-1max_outgoing=-1ports=1,2,3,4context=defaultmsns=*; 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> OUTPUTdev*CLI> dev*CLI> P[ 3] I IND :SETUP oad:6937351046 dad:5157753 pid:19 state:noneP[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0P[ 3] --> Bearer: SpeechP[ 3] --> Codec: AlawP[ 0] --> * NEW CHANNEL dad:5157753 oad:6937351046P[ 3] --> CTON: UnknownP[ 3] --> PRES: Restricted (0)P[ 3] --> SCREEN: Unscreened (0)P[ 3] I SEND:PROCEEDING oad:6937351046 dad:05157753 pid:19P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0P[ 3] --> TRANSPARENT ModeP[ 3] --> incoming_early_audio offP[ 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 ringingP[ 3] * IND : ringing pid:19P[ 3] --> * IND : ringing pid:19P[ 3] I SEND:ALERTING oad:6937351046 dad:05157753 pid:19P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0P[ 3] --> incoming_early_audio offP[ 3] I IND :RELEASE oad:6937351046 dad:05157753 pid:19 state:ALERTINGP[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0P[ 3] --> queue_hangupP[ 3] * RELEASING CHANNEL pid:19 ctx:default dad:s oad:6937351046 state: ALERTINGP[ 3] I SEND:RELEASE_COMPLETE oad:6937351046 dad:05157753 pid:19P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:P[ 3] --> info_dad: onumplan:0 dnumplan:2 rnumplan: cpnnumplan:0P[ 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>