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Thread: need help for isdn4bsd-asterisk setting!
need help for isdn4bsd-asterisk setting!
country flaguser name
China		 2008-03-04 21:13:52 

  


  

    hello, all of users:
i have installed isdn4bsd with Openvox B400P. everything seems ok. but i can not make calls. i am confusing the isdnconfig setting and capi.conf for four port card.
what i did is run:
************************************************
new-host# isdnconfig -u 11 -a -p DRVR_DSS1_TE
new-host# isdnconfig -u 10 -a -p DRVR_DSS1_TE
new-host# isdnconfig -u 9 -a -p DRVR_DSS1_TE
new-host# isdnconfig -u 8 -a -p DRVR_DSS1_TE
new-host# isdnconfig
controller 8 = {
  Layer 1:
    description : HFC-4S PCI ISDN adapter
    type   ;     : passive ISDN (Basic Rate, 2xB)
 ;   channels    : 0x3
   ; serial      : 0xabd5
    power_save  : on
    dialtone    : enabled
    attached    : yes
   ;
 PH-state    : F4: Awaiting signal
  Layer 2:
    driver_type : DRVR_DSS1_TE
}
controller 9 = {
  Layer 1:
    description : HFC-4S PCI ISDN adapter
    type   ;     : passive ISDN (Basic Rate, 2xB)
 ;   channels    : 0x3
   ; serial      : 0xabd6
    power_save  : on
    dialtone    : enabled
    attached    : yes
   ; PH-state    : F3: Deactivated
  Layer 2:
    driver_type : DRVR_DSS1_TE
}
controller 10 = {
  Layer 1:
    description : HFC-4S PCI ISDN adapter
    type   ;     : passive ISDN (Basic Rate, 2xB)
 ;  
 channels    : 0x3
   ; serial      : 0xabd7
    power_save  : on
    dialtone    : enabled
    attached    : yes
   ; PH-state    : F4: Awaiting signal
  Layer 2:
    driver_type : DRVR_DSS1_TE
}
controller 11 = {
  Layer 1:
    description : HFC-4S PCI ISDN adapter
    type   ;     : passive ISDN (Basic Rate, 2xB)
 ;   channels    : 0x3
   ; serial      : 0xabd8
    power_save  : on
    dialtone    : enabled
    attached    : yes
   ; PH-state    : F7: Activated
  Layer
 2:
    driver_type : DRVR_DSS1_TE
}
;**************************************************
; example "capi.conf"
;
; FreeBSD: /usr/local/etc/asterisk/capi.conf
; NetBSD:  /usr/pkg/etc/asterisk/capi.conf
; Linux:   /etc/asterisk/capi.conf
;

[general]
;
; In countries like Norway, the nationalprefix should
; just be left empty.
;
nationalprefix=0
internationalprefix=00
rxgain=1.0
txgain=1.0
;ulaw=yes     ;   ;set this, if you live in u-law world instead of a-law
;debug=yes ;      ;set this, if capi debugging should be enabled by default

; interface sections ...

;
; This is an example for an ISDN adapter
; configured for TE-mode:
;

[ISDN1]      ;    ;this example interface gets name 'ISDN1' and may be
 any
   ;           ;   ;name not starting with 'g' or 'contr'.
isdnmode=msn  ;   ;'MSN' (point-to-multipoint)
incomingmsn=*    ;allow incoming calls to this list of MSNs/DIDs, * == any

;
; Format of "incomingmsn" is like this:
;
; 0) This will only allow any MSN:
;
; incomingmsn=*
;
; 1) This will only allow (MSN == "1"):
;
; incomingmsn=1
;
; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):
;
; incomingmsn=1,2,3
;
; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):
;
; incomingmsn=1*,2,3*
;
; NOTE: When a number matches "1*", everything preceeding the "*" is 
; stripped away from the incoming number. For example if "incomingmsn=1*" and 
; the MSN is 1234, only 234 is passed to Asterisk.
;

controller=0    
 ;ISDN4BSD default (first controller)
group=1      ;    ;dialout group
;prefix=0   ;     ;set a prefix to calling number on incoming calls
softdtmf=on     ;enable/disable software dtmf detection
relaxdtmf=off    ;in addition to softdtmf, you can use 
         ;relaxed dtmf detection, which implies softdtmf=yes
accountcode=     ;Asterisk accountcode to use in CDRs
context=isdn_in_te ;context for incoming calls
holdtype=local   ;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)
echocancel=no    ;disable echo canceller
;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older 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)
;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing 
         ;           ;       ; any audio (outgoing calls in te-mode only)

;dtmf_generate=yes ; set this if your [SIP] phone does not generate
           ; inband DTMF tones. It is not recommended to
           ; enable this. You should configure your [SIP] phone
           ; to generate both inband DTMF and SIP INFO.

;
; This is an example for an ISDN adapter
; configured for NT-mode:
;
[ISDN2]        ;  ;this example interface gets name 'ISDN1' and may be any
   ;           ;   ;name
 not starting with 'g' or 'contr'.
isdnmode=msn  ;   ;'MSN' (point-to-multipoint)
incomingmsn=*    ;allow incoming calls to this list of MSNs/DIDs, * == any

;
; Format of "incomingmsn" is like this:
;
; 0) This will only allow any MSN:
;
; incomingmsn=*
;
; 1) This will only allow (MSN == "1"):
;
; incomingmsn=1
;
; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):
;
; incomingmsn=1,2,3
;
; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):
;
; incomingmsn=1*,2,3*
;
; NOTE: When a number matches "1*", everything preceeding the "*" is 
; stripped away from the incoming number. For example if "incomingmsn=1*" and 
; the MSN is 1234, only 234 is passed to Asterisk.
;

controller=1     ;ISDN4BSD default (first controller)
group=1      ;    ;dialout
 group
;prefix=0   ;     ;set a prefix to calling number on incoming calls
softdtmf=on    ;enable/disable software dtmf detection
relaxdtmf=off    ;in addition to softdtmf, you can use 
         ;relaxed dtmf detection, which implies softdtmf=yes
accountcode=     ;Asterisk accountcode to use in CDRs
context=isdn_in_te ;context for incoming calls
holdtype=local   ;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)
echocancel=no    ;disable echo canceller
;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older 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)
;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing
 
         ;           ;       ; any audio (outgoing calls in te-mode only)

;dtmf_generate=yes ; set this if your [SIP] phone does not generate
           ; inband DTMF tones. It is not recommended to
           ; enable this. You should configure your [SIP] phone
           ; to generate both inband DTMF and SIP INFO.

;
; This is an example for an ISDN adapter
; configured for NT-mode:
;
[ISDN3]        ;  ;this example interface gets name 'ISDN1' and may be any
   ;           ;   ;name not starting with 'g' or 'contr'.
isdnmode=msn  ;   ;'MSN'
 (point-to-multipoint)
incomingmsn=*    ;allow incoming calls to this list of MSNs/DIDs, * == any

;
; Format of "incomingmsn" is like this:
;
; 0) This will only allow any MSN:
;
; incomingmsn=*
;
; 1) This will only allow (MSN == "1"):
;
; incomingmsn=1
;
; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):
;
; incomingmsn=1,2,3
;
; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):
;
; incomingmsn=1*,2,3*
;
; NOTE: When a number matches "1*", everything preceeding the "*" is 
; stripped away from the incoming number. For example if "incomingmsn=1*" and 
; the MSN is 1234, only 234 is passed to Asterisk.
;

controller=2    ;ISDN4BSD default (first controller)
group=1      ;    ;dialout group
;prefix=0   ;     ;set a prefix to calling
 number on incoming calls
softdtmf=on     ;enable/disable software dtmf detection
relaxdtmf=off    ;in addition to softdtmf, you can use 
         ;relaxed dtmf detection, which implies softdtmf=yes
accountcode=     ;Asterisk accountcode to use in CDRs
context=isdn_in_te ;context for incoming calls
holdtype=local   ;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)
echocancel=no    ;disable echo canceller
;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older 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)
;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing
 
         ;           ;       ; any audio (outgoing calls in te-mode only)

;dtmf_generate=yes ; set this if your [SIP] phone does not generate
           ; inband DTMF tones. It is not recommended to
           ; enable this. You should configure your [SIP] phone
           ; to generate both inband DTMF and SIP INFO.

;
; This is an example for an ISDN adapter
; configured for NT-mode:
;
[ISDN4]        ;  ;this example interface gets name 'ISDN1' and may be any
   ;           ;   ;name not starting with 'g' or 'contr'.
isdnmode=msn  ;   ;'MSN'
 (point-to-multipoint)
incomingmsn=*    ;allow incoming calls to this list of MSNs/DIDs, * == any

;
; Format of "incomingmsn" is like this:
;
; 0) This will only allow any MSN:
;
; incomingmsn=*
;
; 1) This will only allow (MSN == "1"):
;
; incomingmsn=1
;
; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):
;
; incomingmsn=1,2,3
;
; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):
;
; incomingmsn=1*,2,3*
;
; NOTE: When a number matches "1*", everything preceeding the "*" is 
; stripped away from the incoming number. For example if "incomingmsn=1*" and 
; the MSN is 1234, only 234 is passed to Asterisk.
;

controller=3     ;ISDN4BSD default (first controller)
group=1      ;    ;dialout group
;prefix=0   ;     ;set a prefix to
 calling number on incoming calls
softdtmf=on     ;enable/disable software dtmf detection
relaxdtmf=off    ;in addition to softdtmf, you can use 
         ;relaxed dtmf detection, which implies softdtmf=yes
accountcode=     ;Asterisk accountcode to use in CDRs
context=isdn_in_te ;context for incoming calls
holdtype=local   ;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)
echocancel=no    ;disable echo canceller
;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older 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)
;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing
 
         ;           ;       ; any audio (outgoing calls in te-mode only)

;dtmf_generate=yes ; set this if your [SIP] phone does not generate
           ; inband DTMF tones. It is not recommended to
           ; enable this. You should configure your [SIP] phone
           ; to generate both inband DTMF and SIP INFO.

;
; This is an example for an ISDN adapter
; configured for NT-mode:
;
*************************************************SIP callout
chan_capi.so => (Common ISDN API 2.0 Driver )
Asterisk Ready.
*CLI>   ;  -- Executing [100from-internal:1] Dial("SIP/600-0871a000", "CAPI/g1/13570807XXX/bl|60") in new stack
  ==
 chan_capi_call:4263:ENTRY=ISDN4:PLCI=0x0003:PBX_CHAN=CAPI/ISDN4/1357080XXXX7:
  ==
    -- Called g1/13570807XXX/bl
[Mar  5 16:01:13] WARNING[698]: chan_capi.c:723 capi_show_conf_error: CAPI: conf_error 0x2003 PLCI=0x00000003 Command=CONNECT_CONF,0x8483
       > CAPI INFO 0x2003: Out of PLCIs
    -- No one is available to answer at this time (1:0/0/0)
  ;  -- Executing [100from-internal:2] Hangup("SIP/600-0871a000", "") in new stack
  == Spawn extension (from-internal, 100, 2) exited non-zero on 'SIP/600-0871a000'
      ; > Out of order update usecount!

********************************
i think, something is wrong in my setting. i google, i could find complete source and instruction for that. Anyone could tell me how to set that for B400P with all TE mode. 
thanks!
James.zhu
 


      
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    Re:  need help for isdn4bsd-asterisk
setting!
  


  

     country flaguser name
Netherlands		
     2008-03-05 04:40:08
  


  

    
On Wed, 2008-03-05 at 11:13 +0800, lizhong zhu wrote:
> hello, all of users:
> i have installed isdn4bsd with Openvox B400P.
everything seems ok. but
> i can not make calls. i am confusing the isdnconfig
setting and
> capi.conf for four port card.

It seems that the 4th port is actually connected. In
capi.conf you have
to name the controller as in isdnconfig.
So
[ISDN1]
controller=8
etc.

Since it seems only controller 11 is connected (4th port) I
suggect that
in [ISDN1], [ISDN2] and [ISDN3] you state group=2.
Only [ISDN4] gets group=1.
When lines are added you can change the group to add that
line to the
dialgroup.

With kind regards,

Pim van Stam
WP van Stam ICT


> what i did is run:
> ************************************************
> new-host# isdnconfig -u 11 -a -p DRVR_DSS1_TE
> new-host# isdnconfig -u 10 -a -p DRVR_DSS1_TE
> new-host# isdnconfig -u 9 -a -p DRVR_DSS1_TE
> new-host# isdnconfig -u 8 -a -p DRVR_DSS1_TE
> new-host# isdnconfig
> controller 8 = {
>   Layer 1:
>     description : HFC-4S PCI ISDN adapter
>     type        : passive ISDN (Basic Rate, 2xB)
>     channels    : 0x3
>     serial      : 0xabd5
>     power_save  : on
>     dialtone    : enabled
>     attached    : yes
>     PH-state    : F4: Awaiting signal
>   Layer 2:
>     driver_type : DRVR_DSS1_TE
> }
> controller 9 = {
>   Layer 1:
>     description : HFC-4S PCI ISDN adapter
>     type        : passive ISDN (Basic Rate, 2xB)
>     channels    : 0x3
>     serial      : 0xabd6
>     power_save  : on
>     dialtone    : enabled
>     attached    : yes
>     PH-state    : F3: Deactivated
>   Layer 2:
>     driver_type : DRVR_DSS1_TE
> }
> controller 10 = {
>   Layer 1:
>     description : HFC-4S PCI ISDN adapter
>     type        : passive ISDN (Basic Rate, 2xB)
>     channels    : 0x3
>     serial      : 0xabd7
>     power_save  : on
>     dialtone    : enabled
>     attached    : yes
>     PH-state    : F4: Awaiting signal
>   Layer 2:
>     driver_type : DRVR_DSS1_TE
> }
> controller 11 = {
>   Layer 1:
>     description : HFC-4S PCI ISDN adapter
>     type        : passive ISDN (Basic Rate, 2xB)
>     channels    : 0x3
>     serial      : 0xabd8
>     power_save  : on
>     dialtone    : enabled
>     attached    : yes
>     PH-state    : F7: Activated
>   Layer 2:
>     driver_type : DRVR_DSS1_TE
> }
> ;**************************************************
> ; example "capi.conf"
> ;
> ; FreeBSD: /usr/local/etc/asterisk/capi.conf
> ; NetBSD:  /usr/pkg/etc/asterisk/capi.conf
> ; Linux:   /etc/asterisk/capi.conf
> ;
> 
> [general]
> ;
> ; In countries like Norway, the nationalprefix should
> ; just be left empty.
> ;
> nationalprefix=0
> internationalprefix=00
> rxgain=1.0
> txgain=1.0
> ;ulaw=yes        ;set this, if you live in u-law world
instead of
> a-law
> ;debug=yes       ;set this, if capi debugging should be
enabled by
> default
> 
> ; interface sections ...
> 
> ;
> ; This is an example for an ISDN adapter
> ; configured for TE-mode:
> ;
> 
> [ISDN1]          ;this example interface gets name
'ISDN1' and may be
> any
>                  ;name not starting with 'g' or
'contr'.
> isdnmode=msn     ;'MSN' (point-to-multipoint)
> incomingmsn=*    ;allow incoming calls to this list of
MSNs/DIDs, * ==
> any
> 
> ;
> ; Format of "incomingmsn" is like this:
> ;
> ; 0) This will only allow any MSN:
> ;
> ; incomingmsn=*
> ;
> ; 1) This will only allow (MSN == "1"):
> ;
> ; incomingmsn=1
> ;
> ; 2) This will only allow (MSN == "1") or
(MSN == "2") or (MSN ==
> "3"):
> ;
> ; incomingmsn=1,2,3
> ;
> ; 3) This will only allow (MSN == "1XX..") or
(MSN == "2") or (MSN ==
> "3XX.."):
> ;
> ; incomingmsn=1*,2,3*
> ;
> ; NOTE: When a number matches "1*",
everything preceeding the "*" is 
> ; stripped away from the incoming number. For example
if
> "incomingmsn=1*" and 
> ; the MSN is 1234, only 234 is passed to Asterisk.
> ;
> 
> controller=0     ;ISDN4BSD default (first controller)
> group=1          ;dialout group
> ;prefix=0        ;set a prefix to calling number on
incoming calls
> softdtmf=on     ;enable/disable software dtmf detection
> relaxdtmf=off    ;in addition to softdtmf, you can use 
>          ;relaxed dtmf detection, which implies
softdtmf=yes
> accountcode=     ;Asterisk accountcode to use in CDRs
> context=isdn_in_te ;context for incoming calls
> holdtype=local   ;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)
> echocancel=no    ;disable echo canceller
> ;echocancelold=yes;use facility selector 6 instead of
correct 8
> (necessary for older 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)
> ;wait_silence_samples=1000 ; wait for 1/8 second of
silence before
> passing 
>                            ; any audio (outgoing calls
in te-mode
> only)
> 
> ;dtmf_generate=yes ; set this if your [SIP] phone does
not generate
>            ; inband DTMF tones. It is not recommended
to
>            ; enable this. You should configure your
[SIP] phone
>            ; to generate both inband DTMF and SIP INFO.
> 
> ;
> ; This is an example for an ISDN adapter
> ; configured for NT-mode:
> ;
> [ISDN2]          ;this example interface gets name
'ISDN1' and may be
> any
>                  ;name not starting with 'g' or
'contr'.
> isdnmode=msn     ;'MSN' (point-to-multipoint)
> incomingmsn=*    ;allow incoming calls to this list of
MSNs/DIDs, * ==
> any
> 
> ;
> ; Format of "incomingmsn" is like this:
> ;
> ; 0) This will only allow any MSN:
> ;
> ; incomingmsn=*
> ;
> ; 1) This will only allow (MSN == "1"):
> ;
> ; incomingmsn=1
> ;
> ; 2) This will only allow (MSN == "1") or
(MSN == "2") or (MSN ==
> "3"):
> ;
> ; incomingmsn=1,2,3
> ;
> ; 3) This will only allow (MSN == "1XX..") or
(MSN == "2") or (MSN ==
> "3XX.."):
> ;
> ; incomingmsn=1*,2,3*
> ;
> ; NOTE: When a number matches "1*",
everything preceeding the "*" is 
> ; stripped away from the incoming number. For example
if
> "incomingmsn=1*" and 
> ; the MSN is 1234, only 234 is passed to Asterisk.
> ;
> 
> controller=1     ;ISDN4BSD default (first controller)
> group=1          ;dialout group
> ;prefix=0        ;set a prefix to calling number on
incoming calls
> softdtmf=on    ;enable/disable software dtmf detection
> relaxdtmf=off    ;in addition to softdtmf, you can use 
>          ;relaxed dtmf detection, which implies
softdtmf=yes
> accountcode=     ;Asterisk accountcode to use in CDRs
> context=isdn_in_te ;context for incoming calls
> holdtype=local   ;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)
> echocancel=no    ;disable echo canceller
> ;echocancelold=yes;use facility selector 6 instead of
correct 8
> (necessary for older 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)
> ;wait_silence_samples=1000 ; wait for 1/8 second of
silence before
> passing 
>                            ; any audio (outgoing calls
in te-mode
> only)
> 
> ;dtmf_generate=yes ; set this if your [SIP] phone does
not generate
>            ; inband DTMF tones. It is not recommended
to
>            ; enable this. You should configure your
[SIP] phone
>            ; to generate both inband DTMF and SIP INFO.
> 
> ;
> ; This is an example for an ISDN adapter
> ; configured for NT-mode:
> ;
> [ISDN3]          ;this example interface gets name
'ISDN1' and may be
> any
>                  ;name not starting with 'g' or
'contr'.
> isdnmode=msn     ;'MSN' (point-to-multipoint)
> incomingmsn=*    ;allow incoming calls to this list of
MSNs/DIDs, * ==
> any
> 
> ;
> ; Format of "incomingmsn" is like this:
> ;
> ; 0) This will only allow any MSN:
> ;
> ; incomingmsn=*
> ;
> ; 1) This will only allow (MSN == "1"):
> ;
> ; incomingmsn=1
> ;
> ; 2) This will only allow (MSN == "1") or
(MSN == "2") or (MSN ==
> "3"):
> ;
> ; incomingmsn=1,2,3
> ;
> ; 3) This will only allow (MSN == "1XX..") or
(MSN == "2") or (MSN ==
> "3XX.."):
> ;
> ; incomingmsn=1*,2,3*
> ;
> ; NOTE: When a number matches "1*",
everything preceeding the "*" is 
> ; stripped away from the incoming number. For example
if
> "incomingmsn=1*" and 
> ; the MSN is 1234, only 234 is passed to Asterisk.
> ;
> 
> controller=2    ;ISDN4BSD default (first controller)
> group=1          ;dialout group
> ;prefix=0        ;set a prefix to calling number on
incoming calls
> softdtmf=on     ;enable/disable software dtmf detection
> relaxdtmf=off    ;in addition to softdtmf, you can use 
>          ;relaxed dtmf detection, which implies
softdtmf=yes
> accountcode=     ;Asterisk accountcode to use in CDRs
> context=isdn_in_te ;context for incoming calls
> holdtype=local   ;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)
> echocancel=no    ;disable echo canceller
> ;echocancelold=yes;use facility selector 6 instead of
correct 8
> (necessary for older 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)
> ;wait_silence_samples=1000 ; wait for 1/8 second of
silence before
> passing 
>                            ; any audio (outgoing calls
in te-mode
> only)
> 
> ;dtmf_generate=yes ; set this if your [SIP] phone does
not generate
>            ; inband DTMF tones. It is not recommended
to
>            ; enable this. You should configure your
[SIP] phone
>            ; to generate both inband DTMF and SIP INFO.
> 
> ;
> ; This is an example for an ISDN adapter
> ; configured for NT-mode:
> ;
> [ISDN4]          ;this example interface gets name
'ISDN1' and may be
> any
>                  ;name not starting with 'g' or
'contr'.
> isdnmode=msn     ;'MSN' (point-to-multipoint)
> incomingmsn=*    ;allow incoming calls to this list of
MSNs/DIDs, * ==
> any
> 
> ;
> ; Format of "incomingmsn" is like this:
> ;
> ; 0) This will only allow any MSN:
> ;
> ; incomingmsn=*
> ;
> ; 1) This will only allow (MSN == "1"):
> ;
> ; incomingmsn=1
> ;
> ; 2) This will only allow (MSN == "1") or
(MSN == "2") or (MSN ==
> "3"):
> ;
> ; incomingmsn=1,2,3
> ;
> ; 3) This will only allow (MSN == "1XX..") or
(MSN == "2") or (MSN ==
> "3XX.."):
> ;
> ; incomingmsn=1*,2,3*
> ;
> ; NOTE: When a number matches "1*",
everything preceeding the "*" is 
> ; stripped away from the incoming number. For example
if
> "incomingmsn=1*" and 
> ; the MSN is 1234, only 234 is passed to Asterisk.
> ;
> 
> controller=3     ;ISDN4BSD default (first controller)
> group=1          ;dialout group
> ;prefix=0        ;set a prefix to calling number on
incoming calls
> softdtmf=on     ;enable/disable software dtmf detection
> relaxdtmf=off    ;in addition to softdtmf, you can use 
>          ;relaxed dtmf detection, which implies
softdtmf=yes
> accountcode=     ;Asterisk accountcode to use in CDRs
> context=isdn_in_te ;context for incoming calls
> holdtype=local   ;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)
> echocancel=no    ;disable echo canceller
> ;echocancelold=yes;use facility selector 6 instead of
correct 8
> (necessary for older 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)
> ;wait_silence_samples=1000 ; wait for 1/8 second of
silence before
> passing 
>                            ; any audio (outgoing calls
in te-mode
> only)
> 
> ;dtmf_generate=yes ; set this if your [SIP] phone does
not generate
>            ; inband DTMF tones. It is not recommended
to
>            ; enable this. You should configure your
[SIP] phone
>            ; to generate both inband DTMF and SIP INFO.
> 
> ;
> ; This is an example for an ISDN adapter
> ; configured for NT-mode:
> ;
> *************************************************SIP
callout
> chan_capi.so => (Common ISDN API 2.0 Driver )
> Asterisk Ready.
> *CLI>     -- Executing [100from-internal:1]
Dial("SIP/600-0871a000",
> "CAPI/g1/13570807XXX/bl|60") in new stack
>   ==
>
chan_capi_call:4263:ENTRY=ISDN4:PLCI=0x0003:PBX_CHAN=CAPI/IS
DN4/1357080XXXX7:
>   ==
>     -- Called g1/13570807XXX/bl
> [Mar  5 16:01:13] WARNING[698]: chan_capi.c:723
capi_show_conf_error:
> CAPI: conf_error 0x2003 PLCI=0x00000003
Command=CONNECT_CONF,0x8483
>        > CAPI INFO 0x2003: Out of PLCIs
>     -- No one is available to answer at this time
(1:0/0/0)
>     -- Executing [100from-internal:2]
Hangup("SIP/600-0871a000", "")
> in new stack
>   == Spawn extension (from-internal, 100, 2) exited
non-zero on
> 'SIP/600-0871a000'
>        > Out of order update usecount!
> 
> ********************************
> i think, something is wrong in my setting. i google, i
could find
> complete source and instruction for that. Anyone could
tell me how to
> set that for B400P with all TE mode. 
> thanks!
> James.zhu
> 
> 
> 
>
____________________________________________________________
__________
>
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> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
> 
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> To UNSUBSCRIBE or update options visit:
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     7.0-RELEASE
  


  

     country flaguser name
United States		
     2008-03-05 05:11:43
  


  

    

  


  

    
Sorry about this gang, but I'm not really a programmer, just and end user of Asterisk. When I rebuilt my server with 7.0-RELEASE and then updated the ports, Asterisk would not install properly, crashing somewhere during the H323 build. So last night I blew the drive away again and started from scratch. This time I disabled the H323 support and Asterisk built without a problem. My phone system is working again but this time without H323.
  
 
  
My question now is isn't H323 an older protocol that was developed for video? What problem can anyone forsee if I don't install the H323 support?
 
      
Be a better friend, newshound, and 
know-it-all with Yahoo! Mobile.  Try it now.

  

  
  
   
     
    

  

    Re:  7.0-RELEASE
  


  

     user name

     2008-03-05 05:47:57
  


  

    
> Sorry about this gang, but I'm not really a programmer,
just and end user of Asterisk. When I rebuilt my server with
7.0-RELEASE and then updated the ports, Asterisk would not
install properly, crashing somewhere during the H323 build.
So last night I blew the drive away again and started from
scratch. This time I disabled the H323 support and Asterisk
built without a problem. My phone system is working again
but this time without H323.
>    
>   My question now is isn't H323 an older protocol that
was developed for video? What problem can anyone forsee if I
don't install the H323 support?

   Yes, H.323 is an older protocol, but it is used for both
audio and video.
The main difference between H.323 and SIP is that H.323 is a
packet-oriented
binary protocol, while SIP is a textual (human readable)
protocol. Most VoIP
hardware these days uses SIP, so you won't have any problems
leaving H.323
support out unless you have some special devices that only
talk H.323.

-DG

Dr. David G. Lawrence
President
Download Technologies, Inc. - http://www.downloadtech.c
om - (866) 399 8500
The FreeBSD Project - http://www.freebsd.org
Pave the road of life with opportunities.

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    Re:  7.0-RELEASE
  


  

     country flaguser name
United Kingdom		
     2008-03-05 05:52:25
  


  

    
> Sorry about this gang, but I'm not really a programmer,
just and end user
> of Asterisk. When I rebuilt my server with 7.0-RELEASE
and then updated
> the ports, Asterisk would not install properly,
crashing somewhere during
> the H323 build. So last night I blew the drive away
again and started from
> scratch. This time I disabled the H323 support and
Asterisk built without
> a problem. My phone system is working again but this
time without H323.
>
>   My question now is isn't H323 an older protocol that
was developed for
> video? What problem can anyone forsee if I don't
install the H323
> support?
>
>
> ---------------------------------
> Be a better friend, newshound, and know-it-all with
Yahoo! Mobile.  Try it
> now._______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--

H323 isn't only for video, it's also used for voice.
Microsoft's
Netmeeting uses it and when you properly setup asterisk with
h323, you can
dial a peer directly from Netmeeting.

I personally don't use H323, but have seen places where
that's all they
had (ie a gatekeeper/mcu/netmeeting setup).

BTW, I have 7.0-STABLE and can build Asterisk from the ports
correctly in
either i386/amd64 on several machines. Perhaps you are
trying to compile
the addons (that which I am also not able too)?

Nicolas.



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