hello, all of users:<br>Firstly, i must say thanks to Hans, and other friends to help me. now it works with OpenVox B400P. It can support LEDs. wow! Since some users have problems with installation, i comes out a more detailed instruction for them. maybe, it can help them. if i missed something in the doc, please add on it. and try to be perfect. in my experience, you have to follow few steps:<br> <div class="MsoNormal" style="text-align: left;" align="left"><span style="font-size: 12pt; font-family: ËÎÌå;" lang="EN-US">hello, all of users:<br> Firstly, i must say thanks to Hans, and other friends to help me. now it works with OpenVox B400P. It can support LEDs. wow! Since some users have problems with installation, i comes out a more detailed instruction for them. maybe, it can help them. if i missed something in the doc, please add on it. and try to be perfect. in my experience, you have to follow few steps:<br> 1) install freebsd-6.2 with source files:<br> run:
sysinstall->configure->distributions->base, kernels, src and sys<br> and ports. if you not sure that, please select more options. it is very important for compiling isdn4bsd and kernels. please check it carefully.<br> 2) portsnap fetch, it will take much time <br> 3) portsnap extract<br> 4) cd to asterisk port(usr/ports/net/asterisk(Asterisk <st1:chsdate year="1899" month="12" day="30" islunardate="False" isrocdate="False" w:st="on">1.4.17</st1:chsdate>) and make install clean to install asterisk with other packages. <br> 5) install Subversion and created one dir call i4b under /usr/src<br> 6) refer the doc from Hans to install svn code<o:p></o:p></span></div> <pre><span lang="EN-US">7) install chan_capi under /usr/src/i4b/i4b/trunk/chan_capi<br><br>8) recompile kernel with capi and i4b. some people use kldload i4b, but i can not do that. edit the GENERIC and add these values:<br><br># i4b required<br><br>option IPR_VJ<br><br>device
"i4bdss1"<br><br>device "i4b"<br><br>device "i4btrc"<br><br>device<br><br><span style=""> </span>"i4bctl"<br><br>device "i4brbch"<br><br>device "i4btel"<br><br>device ihfc<br><br>device sound<br><br>and recompile it and make depend and make install.<br><br>9) after compile the kernel, you have to edit the capi.conf and extensiosn.conf under /usr/local/etc/asterisk. here, i list out my capi.conf and extensiosn.conf as a doc for you<o:p></o:p></span></pre><pre><span lang="EN-US">**************************capi.conf*****************<br><br>; example "capi.conf"<br><br>;<br><br>; FreeBSD: /usr/local/etc/asterisk/capi.conf<br><br>; NetBSD:<span style=""> </span>/usr/pkg/etc/asterisk/capi.conf<br><br>; Linux:<span style=""> </span>/etc/asterisk/capi.conf<br><br>;<br style=""><br><!--[endif]--></span></pre><pre><span lang="EN-US"><br><br>[general]<br><br>;<br><br>; In countries like Norway, the nationalprefix should<br><br>; just be left<br><br><span
style=""> </span>empty.<br><br>;<br><br>nationalprefix=0<br><br>internationalprefix=00<br><br>rxgain=1.0<br><br>txgain=1.0<br><br>;ulaw=yes<span style=""> </span>;set this, if you live in u-law world instead of a-law<br><br>debug=yes<span style=""> </span>;set this, if capi debugging should be enabled by default<br><br><br><br>; interface<br><br><span style=""> </span>sections ...<br><br><br><br>;<br><br>; This is an example for an ISDN adapter<br><br>; configured for TE-mode:<br><br>;<br><br><br><br>[ISDN1]<span style=""> </span>;this example interface gets name 'ISDN1' and may be any<br><br><span style=""> </span>;name not starting with 'g' or 'contr'.<br><br>isdnmode=msn<span style=""> </span>;'MSN'
(point-to-multipoint)<br><br>incomingmsn=*<span style=""> </span>;allow incoming calls to this list of MSNs/DIDs, * == any<br><br><br><br>;<br><br>; Format of "incomingmsn" is like this:<br><br>;<br><br>; 0) This will only allow any MSN:<br><br>;<br><br>; incomingmsn=*<br><br>;<br><br>; 1) This will only allow (MSN == "1"):<br><br>;<br><br>; incomingmsn=1<br><br>;<br><br>; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):<br><br>;<br><br>; incomingmsn=1,2,3<br><br>;<br><br>; 3) This will only allow (MSN == "1XX..") or (MSN<br><br><span style=""> </span>== "2") or (MSN == "3XX.."):<br><br>;<br><br>; incomingmsn=1*,2,3*<br><br>;<br><br>; NOTE: When a number matches "1*", everything preceeding the "*" is <br><br>; stripped away from the incoming number. For example if "incomingmsn=1*" and <br><br>; the MSN is 1234, only 234 is passed<br><br><span style=""> </span>to Asterisk.<br><br>;<br><br><br><br>controller=8<span
style=""> </span>;ISDN4BSD default (first controller)<br><br>group=2<span style=""> </span>;dialout group<br><br>;prefix=0<span style=""> </span>;set a prefix to calling number on incoming calls<br><br>softdtmf=1<span style=""> </span>;enable/disable software dtmf detection<br><br>relaxdtmf=off<span style=""> </span>;in addition to softdtmf, you can use <br><br><span style=""> </span>;relaxed dtmf detection, which implies softdtmf=yes<br><br>accountcode=<span style=""> </span>;Asterisk accountcode to use in CDRs<br><br>context=isdn_in_te ;context for incoming calls<br><br>holdtype=local<span style=""> </span>;when Asterisk puts the call on hold, ISDN HOLD will be used. If<br><br><span style=""> </span><span
style=""> </span>;set to 'local' (default value), no hold is done and Asterisk may<br><br><span style=""> </span>;play MOH.<br><br>immediate=yes<span style=""> </span>;immediate start of pbx with<br><br><span style=""> </span>extension 's' if no digits were<br><br><span style=""> </span>;received on incoming call (no destination number yet)<br><br>echocancel=no<span style=""> </span>;disable echo canceller<br><br>;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)<br><br>;echotail=64<span style=""> </span><br><br><span style=""> </span>;echo cancel tail setting<br><br>bridge=yes<span
style=""> </span>;native bridging (CAPI line interconnect) if available<br><br>;callgroup=1<span style=""> </span>;Asterisk call group<br><br>;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are busy<br><br>devices=2<span style=""> </span>;number of concurrent calls on this controller<br><br><span style=""> </span>;(2 makes sense for single BRI, 30 for PRI)<br><br>;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing <br><br><span style=""> </span>; any audio (outgoing calls in te-mode only)<br><br><br><br>;dtmf_generate=yes ; set this if your [SIP] phone does not generate<br><br><span
style=""> </span>; inband DTMF tones. It is not recommended to<br><br><span style=""> </span>; enable this. You should configure your [SIP] phone<br><br><span style=""> </span>;<br><br><span style=""> </span>to generate both inband DTMF and SIP INFO.<br><br><br><br>;<br><br>; This is an example for an ISDN adapter<br><br>; configured for NT-mode:<br><br>;<br><br>[ISDN2]<span style=""> </span>;this example interface gets name 'ISDN1' and may be any<br><br><span style=""> </span>;name not starting with 'g' or<br><br><span style=""> </span>'contr'.<br><br>isdnmode=msn<span style=""> </span>;'MSN' (point-to-multipoint)<br><br>incomingmsn=*<span style=""> </span>;allow incoming calls to this list of
MSNs/DIDs, * == any<br><br><br><br>;<br><br>; Format of "incomingmsn" is like this:<br><br>;<br><br>; 0) This will only allow any MSN:<br><br>;<br><br>; incomingmsn=*<br><br>;<br><br>; 1) This will only allow (MSN == "1"):<br><br>;<br><br>; incomingmsn=1<br><br>;<br><br>; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):<br><br>;<br><br>; incomingmsn=1,2,3<br><br>;<br><br>; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):<br><br>;<br><br>; incomingmsn=1*,2,3*<br><br>;<br><br>; NOTE: When a number matches "1*", everything preceeding the "*" is <br><br>; stripped away from the incoming number. For example if "incomingmsn=1*" and <br><br>; the MSN<br><br><span style=""> </span>is 1234, only 234 is passed to Asterisk.<br><br>controller=9<span style=""> </span>;ISDN4BSD default (first controller)<br><br>group=2<span style=""> </span>;dialout
group<br><br>;prefix=0<span style=""> </span>;set a prefix to calling number on incoming calls<br><br>softdtmf=0<span style=""> </span>;enable/disable software dtmf<br><br><span style=""> </span>detection<br><br>relaxdtmf=off<span style=""> </span>;in addition to softdtmf, you can use <br><br><span style=""> </span>;relaxed dtmf detection, which implies softdtmf=yes<br><br>accountcode=<span style=""> </span>;Asterisk accountcode to use in CDRs<br><br>context=isdn_in_te ;context for incoming calls<br><br>holdtype=local<span style=""> </span>;when Asterisk puts the call on hold, ISDN HOLD will be used. If<br><br><span style=""> </span>;set to 'local' (default value), no hold is done and Asterisk may<br><br><span
style=""> </span>;play MOH.<br><br>immediate=yes<span style=""> </span>;immediate start of pbx with extension 's' if no digits were<br><br><span style=""> </span>;received on incoming call (no destination number yet)<br><br>echocancel=no<span style=""> </span>;disable echo canceller<br><br>;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older<br><br><span style=""> </span>eicon drivers)<br><br>;echotail=64<span style=""> </span>;echo cancel tail setting<br><br>bridge=yes<span style=""> </span>;native bridging (CAPI line interconnect) if available<br><br>;callgroup=1<span style=""> </span>;Asterisk call group<br><br>;deflect=1234567 ;deflect
incoming calls to 1234567 if all B channels are<br><br><span style=""> </span>busy<br><br>devices=2<span style=""> </span>;number of concurrent calls on this controller<br><br><span style=""> </span>;(2 makes sense for single BRI, 30 for PRI)<br><br>;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing <br><br><span style=""> </span>; any audio (outgoing calls in te-mode only)<br><br><br><br>;dtmf_generate=yes ; set this if your [SIP] phone does not generate<br><br><span style=""> </span>; inband DTMF tones. It is not recommended to<br><br><span style=""> </span>; enable this. You should configure your [SIP]
phone<br><br><span style=""> </span>; to generate both inband DTMF and SIP INFO.<br><br><br><br><span style=""> </span><br><br>[ISDN3]<span style=""> </span>;this example interface gets name 'ISDN1' and may be any<br><br><span style=""> </span>;name not starting with 'g' or 'contr'.<br><br>isdnmode=msn<span style=""> </span>;'MSN'<br><br><span style=""> </span>(point-to-multipoint)<br><br>incomingmsn=*<span style=""> </span>;allow incoming calls to this list of MSNs/DIDs, * == any<br><br><br><br>;<br><br>; Format of "incomingmsn" is like this:<br><br>;<br><br>; 0) This will only allow any MSN:<br><br>;<br><br>; incomingmsn=*<br><br>;<br><br>; 1) This will only allow (MSN == "1"):<br><br>;<br><br>;<br><br><span
style=""> </span>incomingmsn=1<br><br>;<br><br>; 2) This will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):<br><br>;<br><br>; incomingmsn=1,2,3<br><br>;<br><br>; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN == "3XX.."):<br><br>;<br><br>; incomingmsn=1*,2,3*<br><br>;<br><br>; NOTE: When a number matches "1*", everything preceeding the "*" is <br><br>; stripped away from the incoming number. For example if "incomingmsn=1*" and <br><br>; the MSN is 1234, only 234 is passed to Asterisk.<br><br>;<br><br><br><br>controller=10<span style=""> </span>;ISDN4BSD default (first controller)<br><br>group=2<span style=""> </span>;dialout group<br><br>;prefix=0<span style=""> </span>;set a prefix to calling number on incoming calls<br><br>softdtmf=0<span style=""> </span>;enable/disable software dtmf
detection<br><br>relaxdtmf=off<span style=""> </span>;in<br><br><span style=""> </span>addition to softdtmf, you can use <br><br><span style=""> </span>;relaxed dtmf detection, which implies softdtmf=yes<br><br>accountcode=<span style=""> </span>;Asterisk accountcode to use in CDRs<br><br>context=isdn_in_te ;context for incoming calls<br><br>holdtype=local<span style=""> </span>;when Asterisk puts the call on hold, ISDN HOLD will be used.<br><br><span style=""> </span>If<br><br><span style=""> </span>;set to 'local' (default value), no hold is done and Asterisk may<br><br><span style=""> </span>;play MOH.<br><br>immediate=yes<span style=""> </span>;immediate start of pbx with extension 's' if no
digits were<br><br><span style=""> </span>;received on incoming call (no destination number yet)<br><br>echocancel=no<span style=""> </span>;disable echo canceller<br><br>;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)<br><br>;echotail=64<span style=""> </span>;echo cancel tail setting<br><br>bridge=yes<span style=""> </span>;native bridging (CAPI line interconnect) if available<br><br>;callgroup=1<span style=""> </span>;Asterisk call group<br><br>;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are busy<br><br>devices=2<span style=""> </span><span style=""> </span>;number of<br><br><span style=""> </span>concurrent calls on this controller<br><br><span
style=""> </span>;(2 makes sense for single BRI, 30 for PRI)<br><br>;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing <br><br><span style=""> </span>; any audio (outgoing calls in te-mode only)<br><br><br><br>;dtmf_generate=yes ; set this<br><br><span style=""> </span>if your [SIP] phone does not generate<br><br><span style=""> </span>; inband DTMF tones. It is not recommended to<br><br><span style=""> </span>; enable this. You should configure your [SIP] phone<br><br><span style=""> </span>; to generate both inband DTMF and SIP INFO.<br><br><br><br><span style=""> </span><br><br><br><br>[ISDN4]<span
style=""> </span>;this example interface gets name 'ISDN1' and may be any<br><br><span style=""> </span>;name not starting with 'g' or 'contr'.<br><br>isdnmode=msn<span style=""> </span>;'MSN' (point-to-multipoint)<br><br>incomingmsn=*<span style=""> </span>;allow incoming calls to this list of MSNs/DIDs, * == any<br><br><br><br>;<br><br>; Format of "incomingmsn" is like this:<br><br>;<br><br>; 0) This will only allow any MSN:<br><br>;<br><br>; incomingmsn=*<br><br>;<br><br>; 1) This will only allow (MSN == "1"):<br><br>;<br><br>; incomingmsn=1<br><br>;<br><br>; 2) This<br><br><span style=""> </span>will only allow (MSN == "1") or (MSN == "2") or (MSN == "3"):<br><br>;<br><br>; incomingmsn=1,2,3<br><br>;<br><br>; 3) This will only allow (MSN == "1XX..") or (MSN == "2") or (MSN ==
"3XX.."):<br><br>;<br><br>; incomingmsn=1*,2,3*<br><br>;<br><br>; NOTE: When a number matches "1*", everything preceeding the "*" is <br><br>;<br><br><span style=""> </span>stripped away from the incoming number. For example if "incomingmsn=1*" and <br><br>; the MSN is 1234, only 234 is passed to Asterisk.<br><br>;<br><br><br><br>controller=11<span style=""> </span>;ISDN4BSD default (first controller)<br><br>group=1<span style=""> </span>;dialout group<br><br>;prefix=0<span style=""> </span>;set a prefix to calling number on incoming calls<br><br>softdtmf=0<span style=""> </span>;enable/disable software dtmf detection<br><br>relaxdtmf=off<span style=""> </span>;in addition to softdtmf, you can use <br><br><span style=""> </span>;relaxed dtmf detection, which implies softdtmf=yes<br><br>accountcode=<span
style=""> </span>;Asterisk accountcode to use in CDRs<br><br>context=isdn_in_te ;context for incoming calls<br><br>holdtype=local<span style=""> </span>;when Asterisk puts the call on hold, ISDN HOLD will be used. If<br><br><span style=""> </span>;set<br><br><span style=""> </span>to 'local' (default value), no hold is done and Asterisk may<br><br><span style=""> </span>;play MOH.<br><br>immediate=yes<span style=""> </span>;immediate start of pbx with extension 's' if no digits were<br><br><span style=""> </span>;received on incoming call (no destination number yet)<br><br>echocancel=no<span style=""> </span>;disable
echo<br><br><span style=""> </span>canceller<br><br>;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)<br><br>;echotail=64<span style=""> </span>;echo cancel tail setting<br><br>bridge=yes<span style=""> </span>;native bridging (CAPI line interconnect) if available<br><br>;callgroup=1<span style=""> </span>;Asterisk call group<br><br>;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels are busy<br><br>devices=2<span style=""> </span>;number of concurrent calls on this controller<br><br><span style=""> </span>;(2 makes sense for single BRI, 30 for PRI)<br><br>;wait_silence_samples=1000 ; wait for 1/8 second of silence before passing <br><br><span
style=""> </span>; any audio (outgoing calls in te-mode only)<br><br><br><br>;dtmf_generate=yes ; set this if your [SIP] phone does<br><br><span style=""> </span>not generate<br><br><span style=""> </span>; inband DTMF tones. It is not recommended to<br><br><span style=""> </span>; enable this. You should configure your [SIP] phone<br><br><span style=""> </span>; to generate both inband DTMF and SIP INFO.<br><br><br><br>;example "extensions.conf" (Also see "capi.conf" example)<br><br>;<br><br>;<br><br><span style=""> </span>FreeBSD: /usr/local/etc/asterisk/extensions.conf<br><br>; NetBSD:<span style=""> </span>/usr/pkg/etc/asterisk/extensions.conf<br><br>; Linux:<span style="">
</span>/etc/asterisk/extensions.conf<br><br>;<br><br>********************extensions.conf********************<br><br>[isdn_in_te]<br><br>exten<span style=""> </span>=> s,1,NoOp(Invalid incoming call ${EXTEN})<br><br>exten<span style=""> </span>=> s,2,Dial(SIP/600)<br><br>[from-internal]<br><br>exten => 100,1,Dial(CAPI/contr11/1342439XXXXXX/b1,100,Tt)<br><br>exten => 100,2,Hangup<br><br>exten => 500,1,Dial(sip/500)<br><br>exten => 500,2,Hganup<br><br>;<br><br>;exten => 100,1,Dial(CAPI/ISDN1/${CALLNUMBER[${CALLERIDNUM}]}:${DIALSTR}/bl)<br><br>10) run the isdnconfig:<br><br><span style=""> </span>isdnconfig -u 8(9,10,11) pcm_master<span style=""> </span>-p DRVR_DSS1_TE<br><br>11) run asterisk: asterisk -vvvvvvvvvvgc, pleae check chan_capi.so<br><br>i called to controller 11, it works. i can call out and calling using this port.<br><br>it also show erros when inbound call comes
in:<br><br>*************************************<br><br><span style=""> </span>== cd_free:1817:ENTRY=ISDN4:PLCI=0x040b:PBX_CHAN=CAPI/ISDN4/-2:<br><br><span style=""> </span>==<br><br><span style=""> </span>> CAPI: Command=DISCONNECT_IND, 0x<st1:chmetcnv unitname="C" sourcevalue="848" hasspace="False" negative="False" numbertype="1" tcsc="0" w:st="on">848c</st1:chmetcnv>: no call descriptor for PLCI=0x040b, MSGNUM=0x0000:<br><br>[Mar<span style=""> </span>7 15:55:35] ERROR[692]: chan_capi.c:1643 cd_by_pbx_chan: PBX channel has no interface!<br><br><span style=""> </span>== Spawn extension (isdn_in_te, s, 2) exited non-zero on 'CAPI/ISDN4/-2'<br><br>[Mar<span style=""> </span>7 15:55:35] ERROR[692]: chan_capi.c:1643 cd_by_pbx_chan: PBX channel has no interface!<br><br><span style=""> </span>> Data = 'ISDN4/'<br><br><span
style=""> </span>> Out of order update usecount!<br><br>*************************<br><br>that's all i did. hope somebody add more and make it as much complete as<br><br><span style=""> </span>possible.<br><br>Regards,<br><br><st1:PersonName w:st="on">James</st1:PersonName>.zhu from OpenVox<o:p></o:p></span></pre> <div class="MsoNormal" style="text-align: left;" align="left"><span style="font-size: 12pt; font-family: ËÎÌå;" lang="EN-US"><o:p> </o:p></span></div> <div class="MsoNormal" style="text-align: left;" align="left"><span style="font-size: 12pt; font-family: ËÎÌå;" lang="EN-US"><br> <br style=""> <!--[if !supportLineBreakNewLine]--><br style=""> <!--[endif]--><o:p></o:p></span></div> <div class="MsoNormal" style="text-align: left;" align="left"><span style="font-size: 12pt; font-family: ËÎÌå;" lang="EN-US"><o:p> </o:p></span></div> <div class="MsoNormal" style="text-align: left;" align="left"><span
style="font-size: 12pt; font-family: ËÎÌå;" lang="EN-US"><br> <br style=""> <!--[if !supportLineBreakNewLine]--><br style=""> <!--[endif]--><o:p></o:p></span></div> <div class="MsoNormal"><span style="font-size: 12pt;" lang="EN-US"><o:p> </o:p></span></div> <br><br><p> 
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