[Asterisk-bsd] need help for isdn4bsd-asterisk setting!
lizhong zhu
zhulizhongum at yahoo.com.cn
Tue Mar 4 21:13:52 CST 2008
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 [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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