hello, Sir:<br>i tried to edit the capi.conf and dialplan. the test result is this:<br>new-host# capitest -u 11 -o 13424390742<br>main.c: capi_send_listen_request: sending listen request for incoming_calls<br>main.c: capi_send_listen_request: sending listen request for incoming_calls<br>main.c: capi_send_listen_request: sending listen request for incoming_calls<br>main.c: capi_send_listen_request: sending listen request for incoming_calls<br>dialing out, 1 / 1 ...<br>main.c: cd_event: disconnected: normal call clearing<br>************************************************************<br>the message shows this:<br>CAPI_DISCONNECT_RESP {<br> header {<br> WORD wLen = 0x0000<br> WORD wApp = 0x0000<br>
WORD wCmd = 0x8473<br> WORD wNum = 0x0000<br> DWORD dwCid = 0x0000030b<br> }<br> data {<br> }<br>}<br><br> == cd_free:1817:ENTRY=ISDN4:PLCI=0x030b:PBX_CHAN=CAPI/ISDN4/1342439XXX:<br> ==<br> > CAPI: Command=DISCONNECT_IND, 0x848c: no call descriptor for PLCI=0x030b, MSGNUM=0x0000:<br> > Data = 'ISDN4/13424390XXX'<br>[Mar 7 02:40:02] ERROR[659]: chan_capi.c:1643 cd_by_pbx_chan: PBX channel has no interface!<br>[Mar 7 02:40:02] ERROR[659]: chan_capi.c:1643 cd_by_pbx_chan: PBX channel has no
interface!<br> -- No one is available to answer at this time (1:0/0/0)<br> -- Executing [100@from-internal:2] Hangup("SIP/600-08733000", "") in new stack<br> == Spawn extension (from-internal, 100, 2) exited non-zero on 'SIP/600-08733000'<br> > Data = 'ISDN4/1342439XXXX'<br> > Out of order update usecou<br>*****************************************************<br>extensions.conf********************<br>[from-internal]<br>exten => 100,1,Dial(CAPI/contr11/13424390742,100)<br>exten => 100,2,Hangup<br>i call contr11/134XXXXXX<br>i changed to call-limit=10000. i do not think call-limit to control that.<br>any idea for that? thanks<br>James.zhu <br><br><br><b><i>Pim van Stam <pim@vanstam-ict.nl></i></b> дµÀ£º<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> <br>On Wed, 2008-03-05 at
11:13 +0800, lizhong zhu wrote:<br>> hello, all of users:<br>> i have installed isdn4bsd with Openvox B400P. everything seems ok. but<br>> i can not make calls. i am confusing the isdnconfig setting and<br>> capi.conf for four port card.<br><br>It seems that the 4th port is actually connected. In capi.conf you have<br>to name the controller as in isdnconfig.<br>So<br>[ISDN1]<br>controller=8<br>etc.<br><br>Since it seems only controller 11 is connected (4th port) I suggect that<br>in [ISDN1], [ISDN2] and [ISDN3] you state group=2.<br>Only [ISDN4] gets group=1.<br>When lines are added you can change the group to add that line to the<br>dialgroup.<br><br>With kind regards,<br><br>Pim van Stam<br>WP van Stam ICT<br><br><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<br>> a-law<br>> ;debug=yes ;set this, if capi debugging should be enabled by<br>> 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<br>> 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, * ==<br>> 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 ==<br>> "3"):<br>> ;<br>> ; incomingmsn=1,2,3<br>> ;<br>> ; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN ==<br>> "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<br>> "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<br>> be used. If<br>> ;set to 'local' (default value), no hold is done and<br>> Asterisk may<br>> ;play MOH.<br>> immediate=yes ;immediate start of pbx with extension 's' if no<br>> digits were<br>> ;received on incoming call (no destination number<br>> yet)<br>> echocancel=no ;disable echo canceller<br>> ;echocancelold=yes;use facility selector 6 instead of correct 8<br>> (necessary for older eicon drivers)<br>> ;echotail=64 ;echo cancel tail setting<br>> ;bridge=yes ;native bridging (CAPI line interconnect) if<br>>
available<br>> ;callgroup=1 ;Asterisk call group<br>> ;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels<br>> 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<br>> passing <br>> ; any audio (outgoing calls in te-mode<br>> 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<br>> 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, * ==<br>> 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 ==<br>> "3"):<br>> ;<br>> ; incomingmsn=1,2,3<br>> ;<br>> ; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN ==<br>> "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<br>> "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<br>> be used. If<br>> ;set to 'local' (default value), no hold is done and<br>> Asterisk may<br>> ;play MOH.<br>> immediate=yes ;immediate start of pbx with extension 's' if no<br>> digits were<br>> ;received on incoming call (no destination number<br>> yet)<br>> echocancel=no ;disable echo canceller<br>> ;echocancelold=yes;use facility
selector 6 instead of correct 8<br>> (necessary for older eicon drivers)<br>> ;echotail=64 ;echo cancel tail setting<br>> ;bridge=yes ;native bridging (CAPI line interconnect) if<br>> available<br>> ;callgroup=1 ;Asterisk call group<br>> ;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels<br>> 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<br>> passing <br>> ; any audio (outgoing calls in te-mode<br>> 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<br>> 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, * ==<br>> 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 ==<br>> "3"):<br>> ;<br>> ; incomingmsn=1,2,3<br>> ;<br>> ; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN ==<br>> "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<br>> "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<br>> be used. If<br>> ;set to 'local' (default value), no hold is done and<br>> Asterisk may<br>> ;play MOH.<br>> immediate=yes ;immediate start of pbx with extension 's'
if no<br>> digits were<br>> ;received on incoming call (no destination number<br>> yet)<br>> echocancel=no ;disable echo canceller<br>> ;echocancelold=yes;use facility selector 6 instead of correct 8<br>> (necessary for older eicon drivers)<br>> ;echotail=64 ;echo cancel tail setting<br>> ;bridge=yes ;native bridging (CAPI line interconnect) if<br>> available<br>> ;callgroup=1 ;Asterisk call group<br>> ;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels<br>> 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<br>> passing <br>> ; any audio (outgoing calls in te-mode<br>> 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<br>> 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, * ==<br>> 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 ==<br>> "3"):<br>> ;<br>> ; incomingmsn=1,2,3<br>>
;<br>> ; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN ==<br>> "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<br>> "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<br>> be used.
If<br>> ;set to 'local' (default value), no hold is done and<br>> Asterisk may<br>> ;play MOH.<br>> immediate=yes ;immediate start of pbx with extension 's' if no<br>> digits were<br>> ;received on incoming call (no destination number<br>> yet)<br>> echocancel=no ;disable echo canceller<br>> ;echocancelold=yes;use facility selector 6 instead of correct 8<br>> (necessary for older eicon drivers)<br>> ;echotail=64 ;echo cancel tail setting<br>> ;bridge=yes ;native bridging (CAPI line interconnect) if<br>> available<br>> ;callgroup=1 ;Asterisk call group<br>> ;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels<br>> 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<br>>
passing <br>> ; any audio (outgoing calls in te-mode<br>> 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",<br>> "CAPI/g1/13570807XXX/bl|60") in new stack<br>> ==<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:<br>> 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", "")<br>> in new stack<br>> == Spawn extension (from-internal, 100, 2) exited non-zero on<br>> '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<br>> complete source and instruction for that. Anyone could tell me how to<br>> set that for B400P with all TE mode. <br>> thanks!<br>> James.zhu<br>> <br>> <br>> <br>> ______________________________________________________________________<br>> ÑÅ»¢ÓÊÏä´«µÝÐÂÄê×£¸££¬¸öÐԺؿ¨ËÍÇ×Åó£¡ <br>> _______________________________________________<br>> --Bandwidth and
Colocation Provided by http://www.api-digital.com--<br>> <br>> Asterisk-BSD mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-bsd<br><br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by http://www.api-digital.com--<br><br>Asterisk-BSD mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-bsd</blockquote><br><p> 
<hr size=1><a href="http://cn.mail.yahoo.com/gc/index.html?entry=5&souce=mail_mailletter_tagline">ÑÅ»¢ÓÊÏä´«µÝÐÂÄê×£¸££¬¸öÐԺؿ¨ËÍÇ×Åó£¡</a>