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INSTALL
29
INSTALL
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@ -1,28 +1,33 @@
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Modify the Makefile to fit your system, especially the path to the Asterisk include files.
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INSTALL
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=======
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Currently there are some buildtime configuration parameters. You can enable early B3 connects
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if you want the capi channels to come up very soon and hear the indications from your
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local exchange (native capi indications).
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You can also force software dtmf detection/generation to be used.
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Modify the Makefile to fit your system, especially the path to the Asterisk
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include files.
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To build the driver you will need an installed capi system, including header files.
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Currently there are some buildtime configuration parameters:
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- You can switch to Ulaw.
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to build the channel driver type:
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To build the driver you will need an installed capi system, including header
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files.
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To build the channel driver type:
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---------------------------------
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make
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to install:
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To install type:
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----------------
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make install
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to install a sample configuration:
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To install a sample configuration:
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----------------------------------
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make config
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in /etc/asterisk/modules.conf insert the line:
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In /etc/asterisk/modules.conf insert the line:
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load => chan_capi.so
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and in the [global] section:
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chan_capi.so=yes
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HEY, dont forget a trailing newline at the end of modules.conf!!!
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HEY, don't forget a trailing newline at the end of modules.conf!!!
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181
README
181
README
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@ -25,116 +25,131 @@ Frank Sautter, levigo group
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This chan_capi version includes:
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================================
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- multiple controller support
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- Multiple controller support
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- CID,DNID (callling party, called party)
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- CLIR/CLIP
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- supplementary services, CD,HOLD,RETRIEVE,ECT
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- Supplementary services, CD,HOLD,RETRIEVE,ECT
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- DTMF (dependend on card) + software DTMF support
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- early B3 connects (always,success,never)
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- digital audio (what did you think?)
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- incoming/outgoing calls
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- overlap sending (dialtone)
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- E(xplicit) C(all) T(ransfer) (...although it's done implicit .. but dont tell!)
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- tuneable latency ;) you can configure the size of B3 blocks at compile time
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- Early B3 connects (always,success,never)
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- Digital audio (what did you think?)
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- Incoming/outgoing calls
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- Overlap sending (dialtone)
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- E(xplicit) C(all) T(ransfer) (...although it's done implicit-but don't tell!)
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- Tuneable latency: you can configure the size of B3 blocks at compile time
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(in chan_capi_pvt.h, AST_CAPI_MAX_B3_BLOCK_SIZE)
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the default is 160 samples, for non-VoIP use you can tune it down to 130
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- use asterisk's internal dsp functions for dtmf
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- alaw support
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- ulaw support!
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The default is 160 samples, for non-VoIP use you can tune it down to 130
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- Use asterisks internal DSP functions for DTMF
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- Alaw support
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- Ulaw support!
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- Eicon CAPI echo cancelation (echocancel=1)
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- reject call waiting (ACO)
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- Reject call waiting (ACO)
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- DID for Point to Point mode (a.k.a overlap receiving)
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- experimental echo squelching (echosquelch=1)
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- call progress, no need to add ||r to your dialstring anymore
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- rx/tx gains (rxgain=1.0)
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- call deflection on circuitbusy (makefile option) (deflect=12345678)
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- (inter)national dialing prefix (for callerid) configurable in capi.conf
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- CLI command "capi info" shows B channel status
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- capiECT will announce the callerID since it gets lost on most isdn pbxes
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the called party can press # to drop the call
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- audio syncing (timing outgoing dataB3 on incoming dataB3), supposed to fix
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- Call progress, no need to add ||r to your dialstring anymore
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- Rx/Tx gains (rxgain=1.0)
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- Call deflection on circuit busy (deflect=12345678)
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- (Inter)national dialing prefix (for callerid) configurable in capi.conf
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- CLI command "capi info" shows B channel status of chan_capi
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- CapiECT will announce the callerID since it gets lost on most isdn pbxes
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The called party can press # to drop the call
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- Audio syncing (timing outgoing dataB3 on incoming dataB3), supposed to fix
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the DATA_B3_REQ (error = 0x1103) problem
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- catch all MSN (incomingmsn=*)
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- some configuration enhancements (msn=123,124,125 and controller=1,2,3,4)
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- accountcode= added.
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- finally the echo squelching works!
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- callgroup support
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- fixed pipe leak
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- updated to support the new frame->delivery field
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- compiles with latest cvs with a makefile option (LOOK AT THE MAKEFILE)
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- fixed channel name bug in p2p mode
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- added app_capiNoES for disabling the primitive echo suppressor, use this before
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you start recording voicemail or your files may get choppy
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- fixed for latest cvs (AST_MUTEX_DEFINE_STATIC)
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- fixed for latest cvs (asterisk/parking.h -> asterisk/features.h)
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- fixed for latest cvs ast_pthread_create
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- Catch all MSN (incomingmsn=*)
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- Some configuration enhancements (msn=123,124,125 and controller=1,2,3,4)
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- Added accountcode=
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- Echo squelching (echosquelch=1)
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- Callgroup support
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- Fixed pipe leak
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- Updated to support the new frame->delivery field
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- Compiles with different Asterisk versions (automatic build configuration)
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- Fixed channel name bug in PtP mode
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- Added app_capiNoES for disabling the primitive echo suppressor, use this
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before you start recording voicemail or your files may get choppy
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- Added app_capiFax to receive faxes over CAPI (see below)
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- ATTENTION! the dialstring syntax now uses the zaptel dialstring syntax
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it used to be: Dial(CAPI/[@]<outgoingMSN>:[b|B]<destination>)
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It used to be: Dial(CAPI/[@]<outgoingMSN>:[b|B]<destination>)
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now it is: Dial(CAPI/g<group>/[b|B]<destination>)
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or: Dial(CAPI/contr<controller>/[b|B]<destination>)
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Now it is: Dial(CAPI/g<group>/[b|B]<destination>)
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Or: Dial(CAPI/contr<controller>/[b|B]<destination>)
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CLIP/CLIR is now uses the calling presentation of the calling channel, this can
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CLIP/CLIR now uses the calling presentation of the calling channel, which can
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be modified using the CallingPres() application. Use CallinPres(32) for CLIR.
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That is why the msn= param in capi.conf is now obsolete. The callerID is also
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taken from the calling channel.
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- fixes for BSD (Jan Stocker)
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- Fixes for BSD (Jan Stocker)
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Helper applications
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===================
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kapejod says: "No No No, dont use those yet....!" (except maybe HOLD,ECT...)
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Kapejod says: "No No No, don't use those yet....!" (except maybe HOLD,ECT...)
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app_capiCD.c forwards an unanswered call to another phone (does not rely on sservice CD)
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example:
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exten => s,1,Wait,1
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exten => s,2,capiCD,12345678
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app_capiCD.c
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Forwards an unanswered call to another phone (does not rely on sservice CD)
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Example:
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exten => s,1,Wait,1
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exten => s,2,capiCD,12345678
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app_capiHOLD.c puts an answered call on hold, this has nothing to do with asterisk's onhold thingie (music et al)
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after putting a call onhold, never use the Wait application!
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app_capiHOLD.c
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Puts an answered call on hold, this has nothing to do with asterisk's onhold
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thingie (music et al)
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After putting a call onhold, never use the Wait application!
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app_capiRETRIEVE.c gets the holded call back
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app_capiRETRIEVE.c
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Gets the holded call back
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app_capiECT.c explicit call transfer of the holded call (must put call on hold first!)
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example:
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exten => s,1,Answer
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exten => s,2,capiHOLD
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exten => s,3,capiECT,55:50
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will ECT the call to 50 using 55 as the callerid/outgoing msn
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app_capiECT.c
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Explicit call transfer of the call on hold (must put call on hold first!)
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Example:
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exten => s,1,Answer
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exten => s,2,capiHOLD
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exten => s,3,capiECT,55:50
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Will ECT the call to 50 using 55 as the callerid/outgoing MSN
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Using CLIR
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==========
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Use the CallingPres() application before you dial:
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exten => _X.,1,CallingPres(32)
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exten => _X.,2,Dial(CAPI/contr1/${EXTEN})
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exten => _X.,1,CallingPres(32)
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exten => _X.,2,Dial(CAPI/contr1/${EXTEN})
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Enjoying early B3 connects (inband call progress, tones and announcements)
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==========================================================================
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early B3 is now configurable in the dialstring :)
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if you prefix the destination number with a 'b' early B3 will always be used, also if the call fails
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because the number is unprovisioned, etc ...
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if you prefix it with a 'B' early B3 will only be used on successful calls, giving you ring indication,etc...
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Early B3 is now configurable in the dialstring.
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If you prefix the destination number with a 'b' early B3 will always be used,
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also if the call fails, because the number is unprovisioned, etc ...
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If you prefix it with a 'B' early B3 will only be used on successful calls,
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giving you ring indication,etc...
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dont use indications in the Dial command, your local exchange will do that for you:
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exten => _X.,1,Dial(CAPI/contr1/B${EXTEN},30) (early B3 on success)
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exten => _X.,1,Dial(CAPI/contr1/b${EXTEN},30) (always early B3)
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exten => _X.,1,Dial(CAPI/contr1/${EXTEN},30,r) (no early B3, fake ring indication)
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Don't use indications in the Dial command, your local exchange will do that for
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you:
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exten => _X.,1,Dial(CAPI/contr1/B${EXTEN},30)
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(early B3 on success)
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exten => _X.,1,Dial(CAPI/contr1/b${EXTEN},30)
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(always early B3)
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exten => _X.,1,Dial(CAPI/contr1/${EXTEN},30,r)
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(no early B3, fake ring indication)
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exten => _X.,1,Dial(CAPI/contr1/b${EXTEN},30,r) (always early B3, fake indicatons if the exchange
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does not give us indications)
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exten => _X.,1,Dial(CAPI/contr1/B${EXTEN},30,r) (early B3 on success, fake indicatons if the exchange
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does not give us indications)
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exten => _X.,1,Dial(CAPI/contr1/b${EXTEN},30,r)
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(always early B3, fake indicatons if the exchange does not give us
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indications)
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exten => _X.,1,Dial(CAPI/contr1/B${EXTEN},30,r)
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(early B3 on success, fake indicatons if the exchange does not give us
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indications)
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For normal PBX usage you would use the "b" option, always early B3.
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Overlap sending (a.k.a. real dialtone)
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======================================
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when you dial an empty number, and have early B3 enabled, with:
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When you dial an empty number, and have early B3 enabled, with:
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Dial(CAPI/g1/b)
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the channel will come up at once and give you the dialtone it gets from the local exchange.
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at this point the channel is like a legacy phone, now you can send dtmf digits to dial.
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The channel will come up at once and give you the dialtone it gets from the
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local exchange.
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At this point the channel is like a legacy phone, now you can send DTMF digits
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to dial.
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Example context for incoming calls on MSN 12345678:
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===================================================
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@ -143,13 +158,14 @@ Example context for incoming calls on MSN 12345678:
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exten => 12345678,1,Dial(SIP/phone1)
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exten => 12345678,2,Hangup
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short HOWTO of 'capiAnswerFax':
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Short HOWTO of 'capiAnswerFax':
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===============================
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For those of you who have a CAPI card with an on-board DSP (like some Eicon+Diva Server), this patch allows you to receive faxes.
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If you want to answer a channel in fax mode, use capiAnswerFax() instead of+Answer()
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If you use Answer(), you will be in voice mode. If the hardware DSP detects+a fax tone, you can switch from voice to fax mode by calling
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capiAnswerFax().
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For those of you who have a CAPI card with an on-board DSP (like some Eicon and
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DIVA Server), this patch allows you to receive faxes.
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If you want to answer a channel in fax mode, use capiAnswerFax() instead of
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Answer()
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If you use Answer(), you will be in voice mode. If the hardware DSP detects
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fax tone, you can switch from voice to fax mode by calling capiAnswerFax().
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Example of use :
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line number 123, play something, if a fax tone is detected, handle it
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@ -167,12 +183,15 @@ exten => s,2,Hangup()
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exten => h,1,deadagi,fax.php // Run sfftobmp and mail it.
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The output of capiAnswerFax is a SFF file. Use sfftobmp to convert it.
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With a Diva Server, theses features are allowed : fax up to 33600, high
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resolution. Color Fax /JPEG Compression is disabled (I can't test it).
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With a DIVA Server, following features are provided:
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- fax up to 33600
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- high resolution
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- Color Fax
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- JPEG Compression is disabled (I can't test it)
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------
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More information/documentation and commercial support can be found at:
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More information/documentation and commercial support can be found at:
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http://www.junghanns.net/asterisk/
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