<div class="gmail_quote">Thank you very much to all, <br>this configuration works for me too.<br>I use the same enviroment and an openvox B400P.<br><br>Regards,<br><br>Gabriele Ghidelli<br><br><br>On Fri, Feb 26, 2010 at 4:08 AM, qiaoqiao7036 <span dir="ltr"><<a href="mailto:qiaoqiao7036@gmail.com">qiaoqiao7036@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div style="margin: 10px; font-family: verdana; font-size: 10pt;">
<div><font face="Verdana" size="2">Hi,</font></div>
<div> With the help of <i>Armin Schindler </i>i have
installed openvox B200P + mISDN-1.1.9.1 + chan_capi-1.1.4 + asterisk-1.4.26.2
successful.</div>
<div>Thanks! let me show my steps:</div>
<div> </div>
<div>my environment:</div>
<div>intel DG41MJ motherbroad</div>
<div>openvox B200P</div>
<div>debian lenny 5.0.3</div>
<div>mISDN-1.1.9.1</div>
<div>chan_capi-1.1.4</div>
<div>asterisk-1.4.26.2</div>
<div> </div>
<div>1. install the mISDN, asterisk, then modify
chan_capi-1.1.4/chan_capi.c</div>
<div> add some head files:</div>
<div> </div>
<div style="text-indent: 4em;">
<div></div>
<div>#include "chan_capi_platform.h" //under this line</div></div>
<div style="text-indent: 4em;">
<div>#include <asterisk/strings.h></div>
<div>#include <asterisk/dsp.h></div>
<div>#include <asterisk/devicestate.h></div>
<div>#include <asterisk/features.h></div>
<div> </div></div>
<div style="text-indent: 2em;">then complie chan_capi</div>
<div style="text-indent: 2em;"> </div>
<div>2. modify the /usr/sbin/mISDN in line 186, add # in the head, add a new
line " HFCMULTI_layermask[${i}]="0x1f,0x1f" " under line 186</div>
<div> </div>
<div>3. modify /etc/asterisk/modules.conf, add those:</div>
<div>
<div style="text-indent: 2em;">load => res_features.so</div>
<div></div>
<div></div>
<div style="text-indent: 2em;">load => chan_capi.so</div>
<div></div>
<div style="text-indent: 2em;">[global]</div>
<div style="text-indent: 2em;">chan_capi.so=yes</div>
<div style="text-indent: 2em;"> </div>
<div>4. my capi.conf like:</div>
<div> </div>
<div>
<div>[general]</div>
<div>nationalprefix=0 ; or for example "+49"</div>
<div>internationalprefix=00 ; or for example "+"</div>
<div>;subscriberprefix=+4969 ; prefix including area code (some lines need this)</div>
<div>rxgain=1.0 ;linear receive gain (1.0 = no change)</div>
<div>txgain=1.0 ;linear transmit gain (1.0 = no change)</div>
<div>language=de ;set default language</div>
<div>;ulaw=yes ;set this, if you live in u-law world instead of a-law</div>
<div></div>
<div>;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,</div>
<div> ;see Asterisk documentation for all jb* setting available.</div>
<div>;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold.</div>
<div></div>
<div></div>
<div>; interface sections ...</div>
<div></div>
<div>[ISDN1] ;this example interface gets name 'ISDN1' and may be any</div>
<div> ;name not starting with 'g' or 'contr'.</div>
<div> ;Use one interface section for each ISDN port!</div>
<div>;ntmode=yes ;if the ISDN card operates in NT-mode, set this to 'yes'</div>
<div>isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)</div>
<div> ;when using NT-mode, 'DID' should be set in any case</div>
<div>incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * = any</div>
<div>;defaultcid=123 ;set a default caller ID to that interface for dial-out,</div>
<div> ;this caller ID will be used when the dial option 'd' is set.</div>
<div>;controller=0 ;ISDN4BSD default</div>
<div>;controller=7 ;ISDN4BSD USB default</div>
<div>controller=1 ;CAPI controller number of this interface/port</div>
<div>group=1 ;dialout group</div>
<div>;prefix=0 ;set a prefix to the calling number on incoming calls</div>
<div>softdtmf=on ;enable/disable software DTMF detection, recommended for AVM cards</div>
<div>relaxdtmf=on ;in addition to softdtmf, you can use relaxed DTMF detection</div>
<div>faxdetect=off ;enable faxdetection and redirection to EXTEN 'fax' for incoming and/or</div>
<div> ;outgoing calls. (default='off', possible values: 'incoming','outgoing','both')</div>
<div>faxdetecttime=0 ;Only detect faxes during the first 'n' seconds of the call.</div>
<div> ;(default '0' meaning for the whole duration of the call)</div>
<div>;accountcode= ;PBX accountcode to use in CDRs</div>
<div>;amaflags=default;AMA flags for CDR ('default', 'omit', 'billing', or 'documentation')</div>
<div>context=from-isdn ;context for incoming calls</div>
<div>;holdtype=hold ;when the PBX puts the call on hold, ISDN HOLD will be used. If</div>
<div> ;set to 'local' (default value), no hold is done and the PBX may</div>
<div> ;play MOH.</div>
<div>;immediate=yes ;DID: immediate start of PBX with extension 's' if no digits were</div>
<div> ; received on incoming call (no destination number yet)</div>
<div> ;MSN: start PBX on CONNECT_IND and do not wait for SETUP/SENDING-COMPLETE.</div>
<div> ; info like REDIRECTINGNUMBER may be lost, but this is necessary for</div>
<div> ; drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE.</div>
<div>;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression. Disable it before you start recording voicemail</div>
<div> ;or your files may get choppy. (you can use capicommand(echosquelch|no) for this)</div>
<div>;echocancel=yes ;Dialogic(R) Diva(R) (CAPI) echo cancellation (yes=g165)</div>
<div> ;(possible values: 'no', 'yes', 'force', 'g164', 'g165')</div>
<div>;echocancelpath=1;Dialogic(R) Diva(R) (CAPI) echo cancellation path</div>
<div> ;(possible values: default '1' - E.1/T.1/S0, '2' - IP, '3' - both)</div>
<div>;echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)</div>
<div>;echotail=64 ;echo cancel tail setting (default=0 for maximum)</div>
<div>;echocancelnlp=1 ;activate non-linear-processing; this improves echo cancel ratio, but might</div>
<div> ;incorporate variable gain in the signal path.</div>
<div>bridge=no ;native bridging (CAPI line interconnect) if available</div>
<div>;callgroup=1 ;PBX call group</div>
<div>;pickupgroup=1 ;PBX pickup group (which call groups are we allowed to pickup)</div>
<div>;transfergroup=1 ;Controller(s) where a transfer on native bridge is allowed to.</div>
<div>;language=de ;set language for this device (overwrites default language)</div>
<div>;disallow=all ;RTP codec selection (valid with Dialogic(R) Diva(R) Media Boards only)</div>
<div>;allow=all ;RTP codec selection (valid with Dialogic(R) Diva(R) Media Boards only)</div>
<div>devices=2 ;number of concurrent calls (B-Channels) on this controller</div>
<div> ;(2 makes sense for single BRI, 30/23 for PRI/T1)</div>
<div>;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,</div></div>
<div>
<div> ;see Asterisk documentation for all jb* setting available.</div>
<div>;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold.</div>
<div>;qsig=1 ;enable use of Q.SIG extensions. ECMA Variant</div>
<div>;qsig_prnum=1234 ;enable inbound bridging - this should be an QSIG-network-wide unique number</div>
<div></div>
<div>[ISDN2]</div>
<div>;isdnmode=msn</div>
<div>controller=2</div>
<div>bridge=no</div>
<div>echocancel=noi</div>
<div>incomingmsn=*</div>
<div>context=from-isdn</div>
<div>devices=2</div></div>
<div> </div>
<div>5. add those in extensions.conf</div></div>
<div><font face="Verdana" size="2"></font> </div>
<div>
<div>[from-isdn]</div>
<div>exten => s, 1, answer()</div>
<div>exten => s, 2, Dial(SIP/100)</div>
<div>exten => s, 3, hangup</div>
<div></div>
<div>[from-sip]</div>
<div>exten => 1, 1, Dial(CAPI/ISDN2/xxxx(the number you
want to dial)/b)</div>
<div>exten => 1, n, hangup</div>
<div> </div>
<div>6. add a sip account in sip.conf</div>
<div> </div>
<div>
<div>[100]</div>
<div>type=friend</div>
<div>username=100</div>
<div>secret=100</div>
<div>host=dynamic</div>
<div>context=from-sip</div>
<div> </div>
<div>7. Now you can use those command to start:</div>
<div> </div>
<div style="text-indent: 2em;">modprobe capi</div>
<div style="text-indent: 2em;">mISND start</div>
<div style="text-indent: 2em;">asterisk -vvvvvvvvvvvvvvgc</div>
<div style="text-indent: 2em;"> </div>
<div>8. the use command " capi info " & " capi show channels " to
check</div>
<div> </div>
<div>9. You can dial in or dial out with the soft phone if all those step are
work.</div>
<div>
<div>
<div><font face="Verdana" size="2"></font> </div>
<div> </div>
<div> </div>
<div> </div>
<div> </div>
<div> </div>
<div>Reagards</div>
<div>Dennis.Den </div></div></div>
<div> </div></div></div>
<div align="left"><font color="#c0c0c0" face="Verdana" size="2">2010-02-26
</font></div><font face="Verdana" size="2">
<hr style="width: 122px; min-height: 2px;" size="2" align="left">
<div><font color="#c0c0c0" face="Verdana" size="2"><span>qiaoqiao7036</span>
</font></div></font></div>
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<br></blockquote></div><br>