[Asterisk-bsd] 回复： Re: need help for isdn4bsd-asterisk setting!
lizhong zhu
zhulizhongum at yahoo.com.cn
Thu Mar 6 07:57:33 CST 2008
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 [100 at from-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 <pim at vanstam-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
> 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 [100 at from-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 [100 at from-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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