diff options
Diffstat (limited to 'trunk/configs/misdn.conf.sample')
| -rw-r--r-- | trunk/configs/misdn.conf.sample | 455 |
1 files changed, 0 insertions, 455 deletions
diff --git a/trunk/configs/misdn.conf.sample b/trunk/configs/misdn.conf.sample deleted file mode 100644 index 65bdda3ea..000000000 --- a/trunk/configs/misdn.conf.sample +++ /dev/null @@ -1,455 +0,0 @@ -; -; 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 - - -; -; 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. -; - -;------------------------------ 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. - -; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". -;----------------------------------------------------------------------------------- - -[default] - -; define your default context here -; -; default value: default -; -context=misdn - -; language -; -; default value: en -; -language=en - -; -; sets the musiconhold class -; -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 define which bearers should be allowed -; -allowed_bearers=all - -; Prefixes for national and international, those are put before the -; oad if an according dialplan is set by the other end. -; -; default values: nationalprefix : 0 -; internationalprefix : 00 -; -nationalprefix=0 -internationalprefix=00 - -; 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 - - - -; -; 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 - 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 - - -; 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 (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=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 - - -; -; these are the exact isdn screening and presentation indicators -; if -1 is given for both values the presentation indicators are used -; from asterisks SetCallerPres application. -; s=0, p=0 -> callerid presented not screened -; s=1, p=1 -> callerid presented but screened (the remote end does not see it!) -; -; default values s=-1, p=-1 -presentation=-1 -screen=-1 - -; this enables echocancellation, with the given number of taps -; be aware, move this setting only to outgoing portgroups! -; A value of zero turns echocancellation off. -; -; possible values are: 0,32,64,128,256,yes(=128),no(=0) -; -; default value: no -; -;echocancel=no - -; Set this to no to disable echotraining. You can enter a number > 10 -; the value is a multiple of 0.125 ms. -; -; default value: no -; yes = 2000 -; no = 0 -; -echotraining=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 - -[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=15 -msns=102,144,101,104 |