hello, all of users:<br>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.<br>what i did is run:<br>************************************************<br>new-host# isdnconfig -u 11 -a -p DRVR_DSS1_TE<br>new-host# isdnconfig -u 10 -a -p DRVR_DSS1_TE<br>new-host# isdnconfig -u 9 -a -p DRVR_DSS1_TE<br>new-host# isdnconfig -u 8 -a -p DRVR_DSS1_TE<br>new-host# isdnconfig<br>controller 8 = {<br> Layer 1:<br> description : HFC-4S PCI ISDN adapter<br> type : passive ISDN (Basic Rate, 2xB)<br> channels : 0x3<br> serial : 0xabd5<br> power_save : on<br> dialtone : enabled<br> attached : yes<br>
PH-state : F4: Awaiting signal<br> Layer 2:<br> driver_type : DRVR_DSS1_TE<br>}<br>controller 9 = {<br> Layer 1:<br> description : HFC-4S PCI ISDN adapter<br> type : passive ISDN (Basic Rate, 2xB)<br> channels : 0x3<br> serial : 0xabd6<br> power_save : on<br> dialtone : enabled<br> attached : yes<br> PH-state : F3: Deactivated<br> Layer 2:<br> driver_type : DRVR_DSS1_TE<br>}<br>controller 10 = {<br> Layer 1:<br> description : HFC-4S PCI ISDN adapter<br> type : passive ISDN (Basic Rate, 2xB)<br>
channels : 0x3<br> serial : 0xabd7<br> power_save : on<br> dialtone : enabled<br> attached : yes<br> PH-state : F4: Awaiting signal<br> Layer 2:<br> driver_type : DRVR_DSS1_TE<br>}<br>controller 11 = {<br> Layer 1:<br> description : HFC-4S PCI ISDN adapter<br> type : passive ISDN (Basic Rate, 2xB)<br> channels : 0x3<br> serial : 0xabd8<br> power_save : on<br> dialtone : enabled<br> attached : yes<br> PH-state : F7: Activated<br> Layer
2:<br> driver_type : DRVR_DSS1_TE<br>}<br>;**************************************************<br>; example "capi.conf"<br>;<br>; FreeBSD: /usr/local/etc/asterisk/capi.conf<br>; NetBSD: /usr/pkg/etc/asterisk/capi.conf<br>; Linux: /etc/asterisk/capi.conf<br>;<br><br>[general]<br>;<br>; In countries like Norway, the nationalprefix should<br>; just be left empty.<br>;<br>nationalprefix=0<br>internationalprefix=00<br>rxgain=1.0<br>txgain=1.0<br>;ulaw=yes ;set this, if you live in u-law world instead of a-law<br>;debug=yes ;set this, if capi debugging should be enabled by default<br><br>; interface sections ...<br><br>;<br>; This is an example for an ISDN adapter<br>; configured for TE-mode:<br>;<br><br>[ISDN1] ;this example interface gets name 'ISDN1' and may be
any<br> ;name not starting with 'g' or 'contr'.<br>isdnmode=msn ;'MSN' (point-to-multipoint)<br>incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * == any<br><br>;<br>; Format of "incomingmsn" is like this:<br>;<br>; 0) This will only allow any MSN:<br>;<br>; incomingmsn=*<br>;<br>; 1) This will only allow (MSN == "1"):<br>;<br>; incomingmsn=1<br>;<br>; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):<br>;<br>; incomingmsn=1,2,3<br>;<br>; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):<br>;<br>; incomingmsn=1*,2,3*<br>;<br>; NOTE: When a number matches "1*", everything preceeding the "*" is <br>; stripped away from the incoming number. For example if "incomingmsn=1*" and <br>; the MSN is 1234, only 234 is passed to Asterisk.<br>;<br><br>controller=0
;ISDN4BSD default (first controller)<br>group=1 ;dialout group<br>;prefix=0 ;set a prefix to calling number on incoming calls<br>softdtmf=on ;enable/disable software dtmf detection<br>relaxdtmf=off ;in addition to softdtmf, you can use <br> ;relaxed dtmf detection, which implies softdtmf=yes<br>accountcode= ;Asterisk accountcode to use in CDRs<br>context=isdn_in_te ;context for incoming calls<br>holdtype=local ;when Asterisk puts the call on hold, ISDN HOLD will be used. If<br> ;set to 'local' (default value), no hold is done and Asterisk may<br> ;play
MOH.<br>immediate=yes ;immediate start of pbx with extension 's' if no digits were<br> ;received on incoming call (no destination number yet)<br>echocancel=no ;disable echo canceller<br>;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)<br>;echotail=64 ;echo cancel tail setting<br>;bridge=yes ;native bridging (CAPI line interconnect) if available<br>;callgroup=1 ;Asterisk call group<br>;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are busy<br>devices=2 ;number of concurrent calls on this controller<br> ;(2 makes sense for single BRI, 30 for
PRI)<br>;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing <br> ; any audio (outgoing calls in te-mode only)<br><br>;dtmf_generate=yes ; set this if your [SIP] phone does not generate<br> ; inband DTMF tones. It is not recommended to<br> ; enable this. You should configure your [SIP] phone<br> ; to generate both inband DTMF and SIP INFO.<br><br>;<br>; This is an example for an ISDN adapter<br>; configured for NT-mode:<br>;<br>[ISDN2] ;this example interface gets name 'ISDN1' and may be any<br> ;name
not starting with 'g' or 'contr'.<br>isdnmode=msn ;'MSN' (point-to-multipoint)<br>incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * == any<br><br>;<br>; Format of "incomingmsn" is like this:<br>;<br>; 0) This will only allow any MSN:<br>;<br>; incomingmsn=*<br>;<br>; 1) This will only allow (MSN == "1"):<br>;<br>; incomingmsn=1<br>;<br>; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):<br>;<br>; incomingmsn=1,2,3<br>;<br>; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):<br>;<br>; incomingmsn=1*,2,3*<br>;<br>; NOTE: When a number matches "1*", everything preceeding the "*" is <br>; stripped away from the incoming number. For example if "incomingmsn=1*" and <br>; the MSN is 1234, only 234 is passed to Asterisk.<br>;<br><br>controller=1 ;ISDN4BSD default (first controller)<br>group=1 ;dialout
group<br>;prefix=0 ;set a prefix to calling number on incoming calls<br>softdtmf=on ;enable/disable software dtmf detection<br>relaxdtmf=off ;in addition to softdtmf, you can use <br> ;relaxed dtmf detection, which implies softdtmf=yes<br>accountcode= ;Asterisk accountcode to use in CDRs<br>context=isdn_in_te ;context for incoming calls<br>holdtype=local ;when Asterisk puts the call on hold, ISDN HOLD will be used. If<br> ;set to 'local' (default value), no hold is done and Asterisk may<br> ;play MOH.<br>immediate=yes ;immediate start of pbx with extension 's' if no digits
were<br> ;received on incoming call (no destination number yet)<br>echocancel=no ;disable echo canceller<br>;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)<br>;echotail=64 ;echo cancel tail setting<br>;bridge=yes ;native bridging (CAPI line interconnect) if available<br>;callgroup=1 ;Asterisk call group<br>;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are busy<br>devices=2 ;number of concurrent calls on this controller<br> ;(2 makes sense for single BRI, 30 for PRI)<br>;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing
<br> ; any audio (outgoing calls in te-mode only)<br><br>;dtmf_generate=yes ; set this if your [SIP] phone does not generate<br> ; inband DTMF tones. It is not recommended to<br> ; enable this. You should configure your [SIP] phone<br> ; to generate both inband DTMF and SIP INFO.<br><br>;<br>; This is an example for an ISDN adapter<br>; configured for NT-mode:<br>;<br>[ISDN3] ;this example interface gets name 'ISDN1' and may be any<br> ;name not starting with 'g' or 'contr'.<br>isdnmode=msn ;'MSN'
(point-to-multipoint)<br>incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * == any<br><br>;<br>; Format of "incomingmsn" is like this:<br>;<br>; 0) This will only allow any MSN:<br>;<br>; incomingmsn=*<br>;<br>; 1) This will only allow (MSN == "1"):<br>;<br>; incomingmsn=1<br>;<br>; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):<br>;<br>; incomingmsn=1,2,3<br>;<br>; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):<br>;<br>; incomingmsn=1*,2,3*<br>;<br>; NOTE: When a number matches "1*", everything preceeding the "*" is <br>; stripped away from the incoming number. For example if "incomingmsn=1*" and <br>; the MSN is 1234, only 234 is passed to Asterisk.<br>;<br><br>controller=2 ;ISDN4BSD default (first controller)<br>group=1 ;dialout group<br>;prefix=0 ;set a prefix to calling
number on incoming calls<br>softdtmf=on ;enable/disable software dtmf detection<br>relaxdtmf=off ;in addition to softdtmf, you can use <br> ;relaxed dtmf detection, which implies softdtmf=yes<br>accountcode= ;Asterisk accountcode to use in CDRs<br>context=isdn_in_te ;context for incoming calls<br>holdtype=local ;when Asterisk puts the call on hold, ISDN HOLD will be used. If<br> ;set to 'local' (default value), no hold is done and Asterisk may<br> ;play MOH.<br>immediate=yes ;immediate start of pbx with extension 's' if no digits were<br> ;received on incoming
call (no destination number yet)<br>echocancel=no ;disable echo canceller<br>;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)<br>;echotail=64 ;echo cancel tail setting<br>;bridge=yes ;native bridging (CAPI line interconnect) if available<br>;callgroup=1 ;Asterisk call group<br>;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are busy<br>devices=2 ;number of concurrent calls on this controller<br> ;(2 makes sense for single BRI, 30 for PRI)<br>;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing
<br> ; any audio (outgoing calls in te-mode only)<br><br>;dtmf_generate=yes ; set this if your [SIP] phone does not generate<br> ; inband DTMF tones. It is not recommended to<br> ; enable this. You should configure your [SIP] phone<br> ; to generate both inband DTMF and SIP INFO.<br><br>;<br>; This is an example for an ISDN adapter<br>; configured for NT-mode:<br>;<br>[ISDN4] ;this example interface gets name 'ISDN1' and may be any<br> ;name not starting with 'g' or 'contr'.<br>isdnmode=msn ;'MSN'
(point-to-multipoint)<br>incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * == any<br><br>;<br>; Format of "incomingmsn" is like this:<br>;<br>; 0) This will only allow any MSN:<br>;<br>; incomingmsn=*<br>;<br>; 1) This will only allow (MSN == "1"):<br>;<br>; incomingmsn=1<br>;<br>; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):<br>;<br>; incomingmsn=1,2,3<br>;<br>; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):<br>;<br>; incomingmsn=1*,2,3*<br>;<br>; NOTE: When a number matches "1*", everything preceeding the "*" is <br>; stripped away from the incoming number. For example if "incomingmsn=1*" and <br>; the MSN is 1234, only 234 is passed to Asterisk.<br>;<br><br>controller=3 ;ISDN4BSD default (first controller)<br>group=1 ;dialout group<br>;prefix=0 ;set a prefix to
calling number on incoming calls<br>softdtmf=on ;enable/disable software dtmf detection<br>relaxdtmf=off ;in addition to softdtmf, you can use <br> ;relaxed dtmf detection, which implies softdtmf=yes<br>accountcode= ;Asterisk accountcode to use in CDRs<br>context=isdn_in_te ;context for incoming calls<br>holdtype=local ;when Asterisk puts the call on hold, ISDN HOLD will be used. If<br> ;set to 'local' (default value), no hold is done and Asterisk may<br> ;play MOH.<br>immediate=yes ;immediate start of pbx with extension 's' if no digits were<br> ;received on
incoming call (no destination number yet)<br>echocancel=no ;disable echo canceller<br>;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)<br>;echotail=64 ;echo cancel tail setting<br>;bridge=yes ;native bridging (CAPI line interconnect) if available<br>;callgroup=1 ;Asterisk call group<br>;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are busy<br>devices=2 ;number of concurrent calls on this controller<br> ;(2 makes sense for single BRI, 30 for PRI)<br>;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing
<br> ; any audio (outgoing calls in te-mode only)<br><br>;dtmf_generate=yes ; set this if your [SIP] phone does not generate<br> ; inband DTMF tones. It is not recommended to<br> ; enable this. You should configure your [SIP] phone<br> ; to generate both inband DTMF and SIP INFO.<br><br>;<br>; This is an example for an ISDN adapter<br>; configured for NT-mode:<br>;<br>*************************************************SIP callout<br>chan_capi.so => (Common ISDN API 2.0 Driver )<br>Asterisk Ready.<br>*CLI> -- Executing [100@from-internal:1] Dial("SIP/600-0871a000", "CAPI/g1/13570807XXX/bl|60") in new stack<br> ==
chan_capi_call:4263:ENTRY=ISDN4:PLCI=0x0003:PBX_CHAN=CAPI/ISDN4/1357080XXXX7:<br> ==<br> -- Called g1/13570807XXX/bl<br>[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<br> > CAPI INFO 0x2003: Out of PLCIs<br> -- No one is available to answer at this time (1:0/0/0)<br> -- Executing [100@from-internal:2] Hangup("SIP/600-0871a000", "") in new stack<br> == Spawn extension (from-internal, 100, 2) exited non-zero on 'SIP/600-0871a000'<br> > Out of order update usecount!<br><br>********************************<br>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. <br>thanks!<br>James.zhu<br><p> 
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