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Thread: 回复： Re: need help for isdn4bsd-asterisk setting!
回复： Re: need help for isdn4bsd-asterisk setting!
country flaguser name
China		 2008-03-06 07:57:33 

  


  

    hello, Sir:
i tried to edit the capi.conf and dialplan. the test result is this:
new-host# capitest -u 11 -o 13424390742
main.c: capi_send_listen_request: sending listen request for incoming_calls
main.c: capi_send_listen_request: sending listen request for incoming_calls
main.c: capi_send_listen_request: sending listen request for incoming_calls
main.c: capi_send_listen_request: sending listen request for incoming_calls
dialing out, 1 / 1 ...
main.c: cd_event: disconnected: normal call clearing
************************************************************
the message shows this:
CAPI_DISCONNECT_RESP {
 header {
  WORD   ;    wLen   ;           ;  = 0x0000
  WORD   ;    wApp   ;           ;  = 0x0000
 
 WORD   ;    wCmd   ;           ;  = 0x8473
  WORD   ;    wNum   ;           ;  = 0x0000
  DWORD ;     dwCid ;           ;   = 0x0000030b
 }
 data {
 }
}

  == cd_free:1817:ENTRY=ISDN4:PLCI=0x030b:PBX_CHAN=CAPI/ISDN4/1342439XXX:
  ==
     ;  > CAPI: Command=DISCONNECT_IND, 0x848c: no call descriptor for PLCI=0x030b, MSGNUM=0x0000:
   ;    > Data = 'ISDN4/13424390XXX'
[Mar  7 02:40:02] ERROR[659]: chan_capi.c:1643 cd_by_pbx_chan: PBX channel has no interface!
[Mar  7 02:40:02] ERROR[659]: chan_capi.c:1643 cd_by_pbx_chan: PBX channel has no
 interface!
    -- No one is available to answer at this time (1:0/0/0)
  ;  -- Executing [100from-internal:2] Hangup("SIP/600-08733000", "") in new stack
  == Spawn extension (from-internal, 100, 2) exited non-zero on 'SIP/600-08733000'
      ; > Data = 'ISDN4/1342439XXXX'
    ;   > Out of order update usecou
*****************************************************
extensions.conf********************
[from-internal]
exten => 100,1,Dial(CAPI/contr11/13424390742,100)
exten => 100,2,Hangup
i call contr11/134XXXXXX
i changed to call-limit=10000. i do not think call-limit to control that.
any idea for that? thanks
James.zhu 


Pim van Stam <pimvanstam-ict.nl> 写道：
 
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
&gt;     type        : passive ISDN (Basic Rate, 2xB)
>     channels    : 0x3
>     serial      : 0xabd5
>;     power_save  : on
>     dialtone    : enabled
&gt;     attached    : yes
>     PH-state    : F4: Awaiting signal
>;   Layer 2:
>     driver_type : DRVR_DSS1_TE
> }
> controller 9 = {
>   Layer 1:
>     description : HFC-4S PCI ISDN adapter
&gt;     type        : passive ISDN (Basic Rate, 2xB)
>     channels    : 0x3
>     serial      : 0xabd6
>;     power_save  : on
>     dialtone    : enabled
&gt;     attached    : yes
>     PH-state    : F3: Deactivated
>   Layer 2:
>     driver_type : DRVR_DSS1_TE
> }
> controller 10 = {
>   Layer 1:
>  
   description : HFC-4S PCI ISDN adapter
&gt;     type        : passive ISDN (Basic Rate, 2xB)
>     channels    : 0x3
>     serial      : 0xabd7
>;     power_save  : on
>     dialtone    : enabled
&gt;     attached    : yes
>     PH-state    : F4: Awaiting signal
>;   Layer 2:
>     driver_type : DRVR_DSS1_TE
> }
> controller 11 = {
>   Layer 1:
>     description : HFC-4S PCI ISDN adapter
&gt;     type        : passive ISDN (Basic Rate, 2xB)
>     channels    : 0x3
>     serial      : 0xabd8
>;     power_save  : on
>     dialtone    : enabled
&gt;     attached    : yes
>     PH-state    : F7: Activated
>   Layer 2:
>     driver_type : DRVR_DSS1_TE
> }
> ;**************************************************
&gt; ; 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
&gt; 
> ; interface sections ...
> 
> ;
> ; This is an example for an ISDN adapter
&gt; ; configured for TE-mode:
&gt; ;
> 
> [ISDN1]          ;this example interface gets name 'ISDN1' and may be
> any
>                  ;name not starting with 'g' or 'contr'.
&gt; 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*
&gt; ;
> ; 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)
&gt; ;echotail=64     ;echo cancel tail setting
&gt; ;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
&gt; 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
&gt; only)
> 
> ;dtmf_generate=yes ; set this if your [SIP] phone does not generate
&gt;            ; 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
&gt; ; configured for NT-mode:
&gt; ;
> [ISDN2]          ;this example interface gets name 'ISDN1' and may be
> any
>                 
 ;name not starting with 'g' or 'contr'.
&gt; 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*
&gt; ;
> ; 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)
&gt; ;echotail=64     ;echo cancel tail setting
&gt; ;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
&gt; 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
&gt; only)
> 
> ;dtmf_generate=yes ; set this if your [SIP] phone does not generate
&gt;            ; 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
&gt; ; configured for NT-mode:
&gt; ;
> [ISDN3]          ;this example interface gets name 'ISDN1' and may be
> any
>                  ;name not starting with 'g' or 'contr'.
&gt; 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*
&gt; ;
> ; 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)
&gt; ;echotail=64     ;echo cancel tail setting
&gt; ;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
&gt; 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
&gt; only)
> 
> ;dtmf_generate=yes ; set this if your [SIP] phone does not generate
&gt;            ;
 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
&gt; ; configured for NT-mode:
&gt; ;
> [ISDN4]          ;this example interface gets name 'ISDN1' and may be
> any
>                  ;name not starting with 'g' or 'contr'.
&gt; 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*
&gt; ;
> ; 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)
&gt; ;echotail=64     ;echo cancel tail setting
&gt; ;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
&gt; 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
&gt; only)
> 
> ;dtmf_generate=yes ; set this if your [SIP] phone does not generate
&gt;            ; 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
&gt; ; configured for NT-mode:
&gt; ;
> *************************************************SIP callout
&gt; 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'
&gt;        > 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!
&gt; James.zhu
> 
> 
> 
> ______________________________________________________________________
> 雅虎邮箱传递新年祝福鲂院乜ㄋ亲朋！ 
> _______________________________________________
> --Bandwidth and
 Colocation Provided by http://www.api-digital.com--
>; 
> Asterisk-BSD mailing list
> To UNSUBSCRIBE or update options visit:
>;    http://lists.digium.com/mailman/listinfo/asterisk-bsd


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雅虎邮箱传递新年祝福鲂院乜ㄋ亲朋！ 

  

  
  
   
     
    

  

    Re:  鍥炲锛 Re:  need help for
isdn4bsd-asterisk setting!
  


  

     country flaguser name
Norway		
     2008-03-06 10:44:12
  


  

    
Hi,

Did you remember to pass the "pcm_master" option
to isdnconfig when 
configuring your multiport ISDN card ? See "man
isdnconfig" for examples.

You can also try 
running "isdndecode-u 'your_isdn_controller_unit_index'
-i -o -x" in the 
background to see what is happening on the ISDN bus.

1) Try dialing in from external while tracing.
2) Try dialing out from PC while tracing.

--HPS

On Thursday 06 March 2008, lizhong zhu wrote:
> hello, Sir:
> i tried to edit the capi.conf and dialplan. the test
result is this:
> new-host# capitest -u 11 -o 13424390742
> main.c: capi_send_listen_request: sending listen
request for incoming_calls
> main.c: capi_send_listen_request: sending listen
request for incoming_calls
> main.c: capi_send_listen_request: sending listen
request for incoming_calls
> main.c: capi_send_listen_request: sending listen
request for incoming_calls
> dialing out, 1 / 1 ...
> main.c: cd_event: disconnected: normal call clearing
>
************************************************************

> the message shows this:
> CAPI_DISCONNECT_RESP {
>  header {
>   WORD       wLen                = 0x0000
>   WORD       wApp                = 0x0000
>   WORD       wCmd                = 0x8473
>   WORD       wNum                = 0x0000
>   DWORD      dwCid               = 0x0000030b
>  }
>  data {
>  }
> }
>

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om--

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