AW: [Asterisk-Users] Some problem with CAPI support
Sebastian Voss
sevoss at t-online.de
Wed Oct 26 08:22:50 MST 2005
Hi Jörg,
vielen Dank für deine Antwort. Ich denke du meinst die extensions.conf
(bin noch Anfänger, was Asterisk angeht).
Ich habe an der Datei noch nichts verändert.
So sieht sie bei mir so aus:
============================
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk's best guess (strongly recommended).
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
autofallthrough=yes
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on an extensions reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one if its included files, will remain set to the previous value.
;
clearglobalvars=no
;
; If priorityjumping is set to 'yes', then applications that support
; 'jumping' to a different priority based on the result of their
operations
; will do so (this is backwards compatible behavior with pre-1.2
releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a 'j' option in their arguments.
;
priorityjumping=no
;
; You can include other config files, use the #include command (without
the ';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk
configuration files. ;#include "filename.conf"
; The "Globals" category contains global variables that can be
referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
;
; Note the 'g2' in the TRUNK variable above. It specifies which group
(defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to
use in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel (aka. ascending
sequential hunt group).
; G: select the highest-numbered non-busy Zap channel (aka. descending
sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than
last time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than
last time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:pass at provider
;
; Any category other than "General" and "Globals" represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example,
1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are
possible
;
; For example the extension _NXXXXXX would match normal 7 digit
dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension. The priority
; "next" or "n" means the previous priority plus one, regardless of
whether
; the previous priority was associated with the current extension or
not.
; The priority "same" or "s" means the same as the previously specified
; priority, again regardless of whether the previous entry was for the
; same extension. Priorities may be immediately followed by a plus sign
; and another integer to add that amount (most useful with 's' or 'n').
; Priorities may then also have an alias, or label, in
; parenthesis after their name which can be used in goto situations
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. Included contexts are included in the order
; they are listed.
;
;[context]
;exten =>
someexten,priority[+offset][(alias)],application(arg1,arg2,...)
;exten => someexten,priority[+offset][(alias)],application,arg1|arg2...
;
; Timing list for includes is
;
; <time range>|<days of week>|<days of month>|<months>
;
;include => daytime|9:00-17:00|mon-fri|*|*
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern. The most commonly used example is
; of course '9' like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;
;
; Sample entries for extensions.conf
;
;
[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Macro(std-exten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})
[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)
[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325
[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164
[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don't have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
;
; Here are the entries you need to participate in the IAXTEL
; call routing system. Most IAXTEL numbers begin with 1-700, but
; there are exceptions. For more information, and to sign
; up, please go to www.gnophone.com or www.iaxtel.com
;
[iaxtel700]
exten =>
_91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
;
; The SWITCH statement permits a server to share the dialplain with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password at bigserver/local
;
; An "lswitch" is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${EXTEN}@othercontext
;
; An "eswitch" is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
;
; eswitch => IAX2/context@${CURSERVER}
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring
the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press
#, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send
to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If
they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If
they press *, send the user into VoicemailMain
[macro-stdPrivacyexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1
extension-priority)
; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1
extension-priority)`
;
exten => s,1,Dial(${ARG2},20|p) ; Ring
the interface, 20 seconds maximum, call screening option (or use P for
databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press
#, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send
to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If
they press #, return to start
exten => s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to
send this call to a polite "Don't call again" script.
exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to
send this call to a telemarketer torture script.
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If
they press *, send the user into VoicemailMain
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10
seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory
message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some
instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the
congratulations
exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not
up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the
demo"
exten => #,n,Hangup ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest at misery.digium.com/s at default) ; Call
the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo
site
exten => 500,n,Goto(s,6) ; Return to the start over
message.
;
; Create an extension, 600, for evaulating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)
;[mainmenu]
;
; Example "main menu" context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks) ; "Thanks for calling
press 1 for sales, 2 for support, ..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them
comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts) ; "Thanks for calling the sales
department. Press 1 for steve, 2 for..."
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
;
; Extensions like the two below can be used for FWD, Nikotel, sipgate
etc.
; Note that you must have a [sipprovider] section in sip.conf whereas
; the otherprovider.net example does not require such a peer definition
;
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r)
;exten =>
_42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
; Real extensions would go here. Generally you want real extensions to
be 4 or 5
; digits long (although there is no such requirement) and start with a
single
; digit that is fairly large (like 6 or 7) so that you have plenty of
room to
; overlap extensions and menu options without conflict. You can alias
them with
; names, too and use global variables
;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel
hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as
listed
;exten => 6245,n,Voicemail(u6245) ; Voicemail
(unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
;exten => 6245,dial+101,Voicemail(b6245) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time
limit
;exten => 6389,1,Dial(MGCP/aaln/1 at 192.168.0.14)
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}
;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is
something like Zap/2
;exten => mark,1,Goto(6275|1) ; alias mark to 6275
;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
;exten => wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;
; Or a conference room (you'll need to edit meetme.conf to enable this
room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "show applications" at
your
; friendly Asterisk CLI prompt.
;
; 'show application <command>' will show details of how you
; use that particular application in this file, the dial plan.
;
Mit Freundlichen Grüßen
Sebastian Voss
Informatecs Systems
-----Ursprüngliche Nachricht-----
Von: gwynpen [mailto:gwynpen at gmail.com]
Gesendet: Mittwoch, 26. Oktober 2005 14:51
An: 'Sebastian Voss'
Betreff: RE: [Asterisk-Users] Some problem with CAPI support
Hallo Sebastian,
wie sieht Dein dialing plan aus, wird der eingehende Ruf im context demo
auch gematcht?
Grüße
Jörg
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Sebastian Voss
> Sent: Wednesday, October 26, 2005 2:39 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Some problem with CAPI support
>
> Hi,
>
> i have set up asterisk on debian sarge (kernel 2.2.6.8-2)
> with chan_capi and AVM Fritz Card PCI. Asterisk starts up
> fine, but it does not respond to any call on the specified MSN.
>
> I have installed the asterisk sample configuration (make sample)
>
> When i enable capi debugging in the asterisk console (capi
> debug), i get the following output when a call on MSN 22 comes in:
>
> Thank you very much for some suggestions to resolve that
> issue. Sorry for the long post.
>
> ====================SNIP=======================
> *CLI> capi debug
> CAPI Debugging Enabled
> *CLI> capi info
> Contr1: 2 B channels total, 2 B channels free.
> *CLI> CONNECT_IND ID=001 #0x0018 LEN=0037
> Controller/PLCI/NCCI = 0x101
> CIPValue = 0x10
> CalledPartyNumber = <80>22
> CallingPartyNumber = <01 80 2a 2a>10
> CalledPartySubaddress = default
> CallingPartySubaddress = default
> BC = <80 90 a3>
> LLC = default
> HLC = <91 81>
> AdditionalInfo = default
>
> -- CONNECT_IND
> (PLCI=0x101,DID=22,CID=**10,CIP=0x10,CONTROLLER=0x1)
> > ISDN1: msn='22' DNID='22' MSN
> == ISDN1: Incoming call '**10' -> '22'
> INFO_IND ID=001 #0x0019 LEN=0018
> Controller/PLCI/NCCI = 0x101
> InfoNumber = 0x70
> InfoElement = <80>22
>
> INFO_RESP ID=001 #0x0019 LEN=0012
> Controller/PLCI/NCCI = 0x101
>
> -- ISDN1: info element CALLED PARTY NUMBER
> > ISDN1: INFO_IND DID digits not used in this state. INFO_IND
> ID=001 #0x001a LEN=0016
> Controller/PLCI/NCCI = 0x101
> InfoNumber = 0x18
> InfoElement = <89>
>
> INFO_RESP ID=001 #0x001a LEN=0012
> Controller/PLCI/NCCI = 0x101
>
> -- Asterisk Urgent handler
> -- ISDN1: info element CHANNEL IDENTIFICATION 89
> -- Asterisk Urgent handler
> INFO_IND ID=001 #0x001b LEN=0018
> Controller/PLCI/NCCI = 0x101
> InfoNumber = 0x70
> InfoElement = <80>22
>
> INFO_RESP ID=001 #0x001b LEN=0012
> Controller/PLCI/NCCI = 0x101
>
> -- ISDN1: info element CALLED PARTY NUMBER
> > ISDN1: INFO_IND DID digits not used in this state. INFO_IND
> ID=001 #0x001c LEN=0016
> Controller/PLCI/NCCI = 0x101
> InfoNumber = 0x18
> InfoElement = <89>
>
> INFO_RESP ID=001 #0x001c LEN=0012
> Controller/PLCI/NCCI = 0x101
>
> -- ISDN1: info element CHANNEL IDENTIFICATION 89
> -- Asterisk Urgent handler
> DISCONNECT_IND ID=001 #0x001d LEN=0014
> Controller/PLCI/NCCI = 0x101
> Reason = 0x0
>
> DISCONNECT_RESP ID=001 #0x001d LEN=0012
> Controller/PLCI/NCCI = 0x101
>
> == ISDN1: CAPI Hangingup
> == ISDN1: Interface cleanup PLCI=0x101
> -- Asterisk Urgent handler
> ====================SNIP=======================
>
>
>
> Here are some configuration details of my setup:
>
> modules.conf:
> =============
>
> [modules]
> autoload=yes
> noload => pbx_gtkconsole.so
> ;load => pbx_gtkconsole.so
> noload => pbx_kdeconsole.so
> noload => app_intercom.so
> noload => chan_modem.so
> noload => chan_modem_aopen.so
> noload => chan_modem_bestdata.so
> noload => chan_modem_i4l.so
> load => res_musiconhold.so
> noload => chan_alsa.so
> ;noload => chan_oss.so
> load => chan_capi.so
> load => res_features.so
>
> [global]
> ;chan_modem.so=yes
> chan_capi.so=yes
>
>
> capi.conf:
> ==========
>
> [general]
> nationalprefix=0
> internationalprefix=00
> rxgain=0.8
> txgain=0.8
> ;ulaw=yes ;set this, if you live in u-law world
> instead of a-law
>
> [ISDN1] ;this example interface gets name 'ISDN1' and may be
> any
> ;name not starting with 'g' or 'contr'.
> ;ntmode=yes ;if isdn card operates in nt mode, set this to yes
> isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward
> dial)
> ;when using NT-mode, ptp should be set in any case
> incomingmsn=22 ;allow incoming calls to this list of MSNs/DIDs, * ==
> any
> msn=22
> ;controller=0 ;ISDN4BSD default
> ;controller=7 ;ISDN4BSD USB default
> controller=1 ;capi controller number to use
> group=1 ;dialout group
> ;prefix=0 ;set a prefix to calling number on incoming calls
> softdtmf=on ;enable/disable software dtmf detection, recommended
> for AVM ca
> rds
> relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
> detection
> accountcode= ;Asterisk accountcode to use in CDRs
> context=demo ;context for incoming calls
> holdtype=hold ;when Asterisk puts the call on hold, ISDN
> HOLD will be
> used. If
> ;set to 'local' (default value), no hold is
> done and Asterisk may
> ;play MOH.
> immediate=yes ;immediate start of pbx with extension 's' if
> no digits
> were
> ;received on incoming call (no destination
> number yet)
> ;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
> ;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation
> ;(possible values: 'no', 'yes', 'force', 'g164',
> 'g165')
> echocancelold=yes;use facility selector 6 instead of correct
> 8 (necessary for ol der eicon drivers)
> ;echotail=64 ;echo cancel tail setting
> ;bridge=yes ;native bridging (CAPI line interconnect) if
> available
> ;callgroup=1 ;Asterisk call group
> ;deflect=1234567 ;deflect incoming calls to 1234567 if all B
> channels are busy
> devices=2 ;number of concurrent calls on this controller
> ;(2 makes sense for single BRI, 30 for PRI)
>
>
>
>
> Best regards
>
> Sebastian Voss
> Informatecs Systems
>
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