Hallo,
versuche nun schon den ganzen Tag über ne Fritzcard auf ein SIP Fon anzurufen.
Der asterisk hängt an der FritzboxFon über ISDN. Versuche ich nun mit der Nst **2 (anal. Nst der FBF) die Nst 51 (ISDN Nst. der FBF) die SIP Nr. 30 zu erreichen bekomme ich diese Meldung auf dem CLI " == ISDN1: Incoming call`**2`->´51´ und anschliessend ISDN1: CAPI Hangingup und sonst nix. Ich kann aber mit dem SIP Nr. 30 über ISDN die Nst. **2 erreichen.SIP untereinander, SIP nach sipgate und andersrum funzt auch.
SUSE 10.0 / asterisk 1.2.5/ chan_capi_cm 0.6.4/ FritzBoxFon (FBF) 7050
extension.conf
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
nat=yes
writeprotect=no
;
[macro-ruf]
exten => s,1,NoOp(wir sind im Macro ruf gelandet)
exten => s,n,Dial,${ARG1}/${ARG2}|30|r
exten => s,n,Goto,s-${DIALSTATUS}|1
exten => s-BUSY,1,voicemail,b${ARG2}
exten => s-BUSY,n,Hangup
exten => s-NOANSWER,1,voicemail,u${ARG2}
exten => s-NOANSWER,n,Hangup
;
;
[echotest]
exten => 81,1,answer
exten => 81,2,wait,1
exten => 81,3,playback,demo-echotest
exten => 81,4,echo
exten => 81,5,playback,demo-echodone
exten => 81,6,hangup
[mailbox_own]
exten => 88,1,answer
exten => 88,n,wait,1
exten => 88,n,voicemailmain,s${CALLERIDNUM}
exten => 88,n,hangup
[lokal]
; Erreichbarkeit der Nst 30-32 untereinander
exten => _3X,1,NoCDR()
exten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr
;exten => _3X,n,macro,ruf|SIP|${EXTEN}
[isdn-out]
exten => _*.,1,Dial,CAPI/contr1/${EXTEN}
[capi-in]
exten => 51,1,Dial,SIP/30
exten => t,2,Hangup
[sipgate_out]
exten => _0.,1,Dial,SIP/${EXTEN}@xxxxx|45|r
[ankommend]
;exten => 1234,1,Dial,SIP/30&SIP/32|30|r
exten => 1234,1,macro,ruf|SIP|30
;
[default]
include => lokal
include => sipgate_out
include => echotest
include => mailbox_own
include => capi-in
include => isdn-out
capi.conf
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de ;set default language
;ulaw=yes ;set this, if you live in u-law world instead of a-law
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * = any
;defaultcid=123 ;set a default caller id to that interface for dial-out,
;this caller id will be used when dial option 'd' is set.
;controller=0 ;ISDN4BSD default
;controller=7 ;ISDN4BSD USB default
controller=1 ;capi controller number to use
group=1 ;dialout group
;prefix=0 ;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
;holdtype=hold ;when Asterisk puts the call on hold, ISDN HOLD will be used. If
;set to 'local' (default value), no hold is done and Asterisk may
;play MOH.
;immediate=yes ;DID: immediate start of pbx with extension 's' if no digits were
; received on incoming call (no destination number yet)
;MSN: start pbx on CONNECT_IND and don't wait for SETUP/SENDING-COMPLETE.
; info like REDIRECTINGNUMBER may be lost, but this is necessary for
; drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE.
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)
;echotail=64 ;echo cancel tail setting
;bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
;language=de ;set language for this device (overwrites default language)
devices=2 ;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
sip.conf
[general]
context=default ; Default context for incoming calls
; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
Language=de
;--------------------------------------------------------------
; Anmeldedaten VoIP Provider
register => 1234:[email protected]/1234
[1234]
type=peer
username=1234
fromuser=1234
secret=xxxx
host=sipgate.de
fromdomain=sipgate.de
insecure=very
canreinvite=no
nat=yes
disallow=all
allow=ulaw
[sipgate_de_in]
type=peer
fromdomain=sipgate.de
host=sipgate.de
disallow=all
;allow=ulaw
context=ankommend
;-------------------------------------------------------------
;Sip Endgeräte 30-32
;-----------------------------------------------------------------
[30]
callerid=Phone 1 <30>
host=dynamic
domain=192.168.178.22
user=30
secret=geheim
type=friend
mailbox=30
;-------
nat=yes
;-------
canreinvite=no
[31]
callerid=Phone 2 <31>
host=dynamic
domain=192.168.178.22
user=31
secret=geheim
type=friend
mailbox=31
;-------
nat=yes
;-------
canreinvite=no
[32]
callerid=Phone 3 <32>
host=dynamic
domain=192.168.178.22
user=32
secret=geheim
type=friend
mailbox=32
;-------
nat=yes
;-------
canreinvite=no
versuche nun schon den ganzen Tag über ne Fritzcard auf ein SIP Fon anzurufen.
Der asterisk hängt an der FritzboxFon über ISDN. Versuche ich nun mit der Nst **2 (anal. Nst der FBF) die Nst 51 (ISDN Nst. der FBF) die SIP Nr. 30 zu erreichen bekomme ich diese Meldung auf dem CLI " == ISDN1: Incoming call`**2`->´51´ und anschliessend ISDN1: CAPI Hangingup und sonst nix. Ich kann aber mit dem SIP Nr. 30 über ISDN die Nst. **2 erreichen.SIP untereinander, SIP nach sipgate und andersrum funzt auch.
SUSE 10.0 / asterisk 1.2.5/ chan_capi_cm 0.6.4/ FritzBoxFon (FBF) 7050
extension.conf
;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the "extensions reload" command in the CLI
; - With the "reload" command (that reloads everything) in the CLI
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
nat=yes
writeprotect=no
;
[macro-ruf]
exten => s,1,NoOp(wir sind im Macro ruf gelandet)
exten => s,n,Dial,${ARG1}/${ARG2}|30|r
exten => s,n,Goto,s-${DIALSTATUS}|1
exten => s-BUSY,1,voicemail,b${ARG2}
exten => s-BUSY,n,Hangup
exten => s-NOANSWER,1,voicemail,u${ARG2}
exten => s-NOANSWER,n,Hangup
;
;
[echotest]
exten => 81,1,answer
exten => 81,2,wait,1
exten => 81,3,playback,demo-echotest
exten => 81,4,echo
exten => 81,5,playback,demo-echodone
exten => 81,6,hangup
[mailbox_own]
exten => 88,1,answer
exten => 88,n,wait,1
exten => 88,n,voicemailmain,s${CALLERIDNUM}
exten => 88,n,hangup
[lokal]
; Erreichbarkeit der Nst 30-32 untereinander
exten => _3X,1,NoCDR()
exten => _3X,n,Dial,SIP/${EXTEN}|55|Ttr
;exten => _3X,n,macro,ruf|SIP|${EXTEN}
[isdn-out]
exten => _*.,1,Dial,CAPI/contr1/${EXTEN}
[capi-in]
exten => 51,1,Dial,SIP/30
exten => t,2,Hangup
[sipgate_out]
exten => _0.,1,Dial,SIP/${EXTEN}@xxxxx|45|r
[ankommend]
;exten => 1234,1,Dial,SIP/30&SIP/32|30|r
exten => 1234,1,macro,ruf|SIP|30
;
[default]
include => lokal
include => sipgate_out
include => echotest
include => mailbox_own
include => capi-in
include => isdn-out
capi.conf
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de ;set default language
;ulaw=yes ;set this, if you live in u-law world instead of a-law
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs, * = any
;defaultcid=123 ;set a default caller id to that interface for dial-out,
;this caller id will be used when dial option 'd' is set.
;controller=0 ;ISDN4BSD default
;controller=7 ;ISDN4BSD USB default
controller=1 ;capi controller number to use
group=1 ;dialout group
;prefix=0 ;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf detection
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
;holdtype=hold ;when Asterisk puts the call on hold, ISDN HOLD will be used. If
;set to 'local' (default value), no hold is done and Asterisk may
;play MOH.
;immediate=yes ;DID: immediate start of pbx with extension 's' if no digits were
; received on incoming call (no destination number yet)
;MSN: start pbx on CONNECT_IND and don't wait for SETUP/SENDING-COMPLETE.
; info like REDIRECTINGNUMBER may be lost, but this is necessary for
; drivers/pbx/telco which does not send SETUP or SENDING-COMPLETE.
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8 (necessary for older eicon drivers)
;echotail=64 ;echo cancel tail setting
;bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
;language=de ;set language for this device (overwrites default language)
devices=2 ;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
sip.conf
[general]
context=default ; Default context for incoming calls
; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
Language=de
;--------------------------------------------------------------
; Anmeldedaten VoIP Provider
register => 1234:[email protected]/1234
[1234]
type=peer
username=1234
fromuser=1234
secret=xxxx
host=sipgate.de
fromdomain=sipgate.de
insecure=very
canreinvite=no
nat=yes
disallow=all
allow=ulaw
[sipgate_de_in]
type=peer
fromdomain=sipgate.de
host=sipgate.de
disallow=all
;allow=ulaw
context=ankommend
;-------------------------------------------------------------
;Sip Endgeräte 30-32
;-----------------------------------------------------------------
[30]
callerid=Phone 1 <30>
host=dynamic
domain=192.168.178.22
user=30
secret=geheim
type=friend
mailbox=30
;-------
nat=yes
;-------
canreinvite=no
[31]
callerid=Phone 2 <31>
host=dynamic
domain=192.168.178.22
user=31
secret=geheim
type=friend
mailbox=31
;-------
nat=yes
;-------
canreinvite=no
[32]
callerid=Phone 3 <32>
host=dynamic
domain=192.168.178.22
user=32
secret=geheim
type=friend
mailbox=32
;-------
nat=yes
;-------
canreinvite=no
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