[Asterisk-bsd] 回复： Re: need help for isdn4bsd-asterisk setting!
Pim van Stam
pim at vanstam-ict.nl
Thu Mar 6 04:19:53 CST 2008
Hi,
You can test if the lines are working with:
# capitest -u 8 -o 'telnr'
# capitest -u 9 -o 'telnr'
# capitest -u 10 -o 'telnr'
# capitest -u 11 -o 'telnr'
Your active line is in unit 11 in stead of 8. You should modify
capi.conf with [ISDN1] group=2 and [ISDN4] group=1
Than dial with CAPI/g1/'number'. With this you are more flexible in
adding lines.
Your problem however lies in sip.conf, I guess. Do you have 'call-limit'
in this? Like:
sip.conf
[1000]
call-limit=50
With kind regards,
Pim van Stam
lizhong zhu wrote:
> hello, all of you:
> i changed to all different setting, but i am not lucky, i still can not make calls.
> ************************ capi.conf for B400P********************
> ;
> ; 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=8 ;ISDN4BSD default (first controller)
> group=1 ;dialout group
> ;prefix=0 ;set a prefix to calling number on incoming calls
> softdtmf=1 ;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=9 ;ISDN4BSD default (first controller)
> group=2 ;dialout group
> ;prefix=0 ;set a prefix to calling number on incoming calls
> softdtmf=0 ;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.
>
>
> [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=10 ;ISDN4BSD default (first controller)
> group=2 ;dialout group
> ;prefix=0 ;set a prefix to calling number on incoming calls
> softdtmf=0 ;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.
>
>
>
> [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=11 ;ISDN4BSD default (first controller)
> group=2 ;dialout group
> ;prefix=0 ;set a prefix to calling number on incoming calls
> softdtmf=0 ;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.
> *****************isdnconfig******************************************
> 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
>
> ; example "extensions.conf" (Also see "capi.conf" example)
> ;
> ; FreeBSD: /usr/local/etc/asterisk/extensions.conf
> ; NetBSD: /usr/pkg/etc/asterisk/extensions.conf
> ; Linux: /etc/asterisk/extensions.conf
> ;
> **********************extensions.conf***************
>
> [isdn_in_te]
> exten => s,1,NoOp(Invalid incoming call ${EXTEN})
> exten => s,2,Goto(isdn_in,${EXTEN},1)
> [from-internal]
> exten => 100,1, Dial(CAPI/ISDN4/82535095)
> exten => 100,2,Hangup
> exten => 500,1,Dial(sip/500)
> exten => 500,2,Hganup
> ;
> ;exten => 100,1,Dial(CAPI/ISDN1/${CALLNUMBER[${CALLERIDNUM}]}:${DIALSTR}/bl)
> ***********************************console tips**********************
> -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 600
>
> *CLI> -- Executing [100 at from-internal:1] Dial("SIP/600-08734000", "CAPI/ISDN4/82535095") in new stack
> == chan_capi_call:4263:ENTRY=ISDN4:PLCI=0x000b:PBX_CHAN=CAPI/ISDN4/82535095:
> ==
> -- Called ISDN4/82535095
> > Out of order update usecount!
> -- No one is available to answer at this time (1:0/0/0)
> -- Executing [100 at from-internal:2] Hangup("SIP/600-08734000", "") in new stack
> == Spawn extension (from-internal, 100, 2) exited non-zero on 'SIP/600-08734000'
> > Out of order update usecount!
> *****************************end*********************
> i tried to make outbound call and inbound calls, it does not work. the LEDs are not on(svn code). Anyone can tell me the details about capi.conf and controller?
> Regards,
> 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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>
>
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