Hi@All, bin neu hier und benötige mal Fachliche Unterstützung 
Es geht um Asterisk Version 1.4 in Verbindung mit einer Eicon Diva Karte.
Linux version 2.6.8-3-386. XLite zum Testen auf zwei Clients installiert.
Das telefonieren über SIP von XLite Softphone zu Softphone funktioniert einwandfrei.
Das Anrufen vom XLite Softphone über ISDN zu einem ISDN Telefon an unserer TK-Anlage über die (Vorwahl + Rufnummer) funktioniert ebenfalls.
Was nicht funktioniert ist der Anruf vom XLite Softphone zu einem ISDN Telefon im Haus, dass z.B. die Durchwahl 524 hat. Ich kann zwar die 524 wählen, im Debugmodus von Asterisk (Asterisk -vvvvvc) wird auch kein Fehler angezeigt, jedoch Klingelt es bei meinen ISDN Telefon komischerweise nicht.
Wenn ich die 524 erreichen will, muss ich über Vorwahl + Rufnummer gehen.
Angeschlossen ist Asterisk an dem S2M Primärmultiplexanschluss.
Anbei poste ich am besten mal meine aktuellen Configs:
/etc/asterisk/capi.conf
----------------------
[general]
nationalprefix=0
internationalprefix=00
rxgain=1.0 ;linear receive gain (1.0 = no change)
txgain=1.0 ;linear transmit gain (1.0 = no change)
language=de ;set default language
;ulaw=yes ;set this, if you live in u-law world instead of a-law
;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold.
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;Use one interface section for each isdn port!
;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 of this interface/port
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
faxdetect=off ;enable faxdetection and redirection to EXTEN 'fax' for incoming and/or
;outgoing calls. (default='off', possible values: 'incoming','outgoing','both')
accountcode= ;PBX accountcode to use in CDRs
;amaflags=default;AMA flags for CDR ('default', 'omit', 'billing', or 'documentation')
context=capi-in ;context for incoming calls
;holdtype=hold ;when the PBX puts the call on hold, ISDN HOLD will be used. If
;set to 'local' (default value), no hold is done and the PBX 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 (yes=g165)
;(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 (default=0 for maximum)
;echocancelnlp=1 ;activate non-linear-processing; this improves echo cancel ratio, but might
;incorporate variable gain in the signal path.
;bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;PBX call group
;pickupgroup=1 ;PBX pickup group (which call groups are we allowed to pickup)
;language=de ;set language for this device (overwrites default language)
;disallow=all ;RTP codec selection (valid with Eicon DIVA Server only)
;allow=all ;RTP codec selection (valid with Eicon DIVA Server only)
devices=2 ;number of concurrent calls (b-channels) on this controller
;(2 makes sense for single BRI, 30/23 for PRI/T1)
;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold.
/etc/asterisk/extensions.conf
----------------------------
[general]
static = yes
writeprotect = no
[local-sip]
exten = 66,1,Dial(SIP/test.adam,8)
exten = 67,2,Dial(SIP/test.christoph,8)
exten = 68,2,Dial(SIP/test.moritz,8)
exten = _XX.,1,Dial(CAPI/ISDN1/${EXTEN},8,tr)
[capi-in]
exten=>590,1,Dial(SIP/test.adam,15,tr)
exten=>590,2,Hangup
exten=>590,102,Hangup
/etc/asterisk/modules.conf
--------------------------
[modules]
autoload=yes
;
load => chan_capi.so
;
[global]
chan_capi.so=yes
/etc/asterisk/sip.conf
---------------------
[general]
port = 5060
bindaddr = 0.0.0.0
;context = sip-in
[test.adam]
type = friend
username = test.adam
secret = geheim
host = dynamic
context = local-sip
regexten = 66
[test.christoph]
type = friend
username = test.christoph
secret = geheim
host = dynamic
context = local-sip
regexten = 67
[test.moritz]
type = friend
username = test.moritz
secret = geheim
host = dynamic
context = local-sip
regexten = 68
Hier noch ein kleiner Auszug aus dem DEBUGMODUS (Asterisk -vvvvvc)
-------------------------------------------------------------------
*CLI> -- Registered SIP 'test.adam' at 192.168.111.107 port 49091 expires 3600
-- Saved useragent "X-Lite release 1006e stamp 34025" for peer test.adam
-- Executing [524@local-sip:1] Dial("SIP/test.adam-081db5e0", "CAPI/ISDN1/524|8|tr") in new stack
-- Called ISDN1/524
-- CAPI/ISDN1/524-0 is making progress passing it to SIP/test.adam-081db5e0
-- Nobody picked up in 8000 ms
== ISDN1#02: CAPI Hangingup for PLCI=0x901 in state 5
> ISDN1#02: CAPI INFO 0x3490: Normal call clearing
[Feb 27 17:17:30] WARNING[4197]: pbx.c:2460 __ast_pbx_run: Timeout, but no rule 't' in context 'local-sip'
//Das gibt er aus wenn ich versuche die 524 über ISDN zu erreichen.
Über jegliche Ünterstützung würde ich mich jedenfalls sehr freuen.
:noidea:
Es geht um Asterisk Version 1.4 in Verbindung mit einer Eicon Diva Karte.
Linux version 2.6.8-3-386. XLite zum Testen auf zwei Clients installiert.
Das telefonieren über SIP von XLite Softphone zu Softphone funktioniert einwandfrei.
Das Anrufen vom XLite Softphone über ISDN zu einem ISDN Telefon an unserer TK-Anlage über die (Vorwahl + Rufnummer) funktioniert ebenfalls.
Was nicht funktioniert ist der Anruf vom XLite Softphone zu einem ISDN Telefon im Haus, dass z.B. die Durchwahl 524 hat. Ich kann zwar die 524 wählen, im Debugmodus von Asterisk (Asterisk -vvvvvc) wird auch kein Fehler angezeigt, jedoch Klingelt es bei meinen ISDN Telefon komischerweise nicht.
Wenn ich die 524 erreichen will, muss ich über Vorwahl + Rufnummer gehen.
Angeschlossen ist Asterisk an dem S2M Primärmultiplexanschluss.
Anbei poste ich am besten mal meine aktuellen Configs:
/etc/asterisk/capi.conf
----------------------
[general]
nationalprefix=0
internationalprefix=00
rxgain=1.0 ;linear receive gain (1.0 = no change)
txgain=1.0 ;linear transmit gain (1.0 = no change)
language=de ;set default language
;ulaw=yes ;set this, if you live in u-law world instead of a-law
;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold.
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;Use one interface section for each isdn port!
;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 of this interface/port
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
faxdetect=off ;enable faxdetection and redirection to EXTEN 'fax' for incoming and/or
;outgoing calls. (default='off', possible values: 'incoming','outgoing','both')
accountcode= ;PBX accountcode to use in CDRs
;amaflags=default;AMA flags for CDR ('default', 'omit', 'billing', or 'documentation')
context=capi-in ;context for incoming calls
;holdtype=hold ;when the PBX puts the call on hold, ISDN HOLD will be used. If
;set to 'local' (default value), no hold is done and the PBX 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 (yes=g165)
;(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 (default=0 for maximum)
;echocancelnlp=1 ;activate non-linear-processing; this improves echo cancel ratio, but might
;incorporate variable gain in the signal path.
;bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;PBX call group
;pickupgroup=1 ;PBX pickup group (which call groups are we allowed to pickup)
;language=de ;set language for this device (overwrites default language)
;disallow=all ;RTP codec selection (valid with Eicon DIVA Server only)
;allow=all ;RTP codec selection (valid with Eicon DIVA Server only)
devices=2 ;number of concurrent calls (b-channels) on this controller
;(2 makes sense for single BRI, 30/23 for PRI/T1)
;jb..... ;with Asterisk 1.4 you can configure jitterbuffer,
;see Asterisk documentation for all jb* setting available.
;mohinterpret=default ;Asterisk 1.4: default music on hold class when placed on hold.
/etc/asterisk/extensions.conf
----------------------------
[general]
static = yes
writeprotect = no
[local-sip]
exten = 66,1,Dial(SIP/test.adam,8)
exten = 67,2,Dial(SIP/test.christoph,8)
exten = 68,2,Dial(SIP/test.moritz,8)
exten = _XX.,1,Dial(CAPI/ISDN1/${EXTEN},8,tr)
[capi-in]
exten=>590,1,Dial(SIP/test.adam,15,tr)
exten=>590,2,Hangup
exten=>590,102,Hangup
/etc/asterisk/modules.conf
--------------------------
[modules]
autoload=yes
;
load => chan_capi.so
;
[global]
chan_capi.so=yes
/etc/asterisk/sip.conf
---------------------
[general]
port = 5060
bindaddr = 0.0.0.0
;context = sip-in
[test.adam]
type = friend
username = test.adam
secret = geheim
host = dynamic
context = local-sip
regexten = 66
[test.christoph]
type = friend
username = test.christoph
secret = geheim
host = dynamic
context = local-sip
regexten = 67
[test.moritz]
type = friend
username = test.moritz
secret = geheim
host = dynamic
context = local-sip
regexten = 68
Hier noch ein kleiner Auszug aus dem DEBUGMODUS (Asterisk -vvvvvc)
-------------------------------------------------------------------
*CLI> -- Registered SIP 'test.adam' at 192.168.111.107 port 49091 expires 3600
-- Saved useragent "X-Lite release 1006e stamp 34025" for peer test.adam
-- Executing [524@local-sip:1] Dial("SIP/test.adam-081db5e0", "CAPI/ISDN1/524|8|tr") in new stack
-- Called ISDN1/524
-- CAPI/ISDN1/524-0 is making progress passing it to SIP/test.adam-081db5e0
-- Nobody picked up in 8000 ms
== ISDN1#02: CAPI Hangingup for PLCI=0x901 in state 5
> ISDN1#02: CAPI INFO 0x3490: Normal call clearing
[Feb 27 17:17:30] WARNING[4197]: pbx.c:2460 __ast_pbx_run: Timeout, but no rule 't' in context 'local-sip'
//Das gibt er aus wenn ich versuche die 524 über ISDN zu erreichen.
Über jegliche Ünterstützung würde ich mich jedenfalls sehr freuen.
:noidea:
