Hello everyone,
I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz,
256MB) with a TDM40b a TDM04b and an avm fritz!card pci.
I experience a problem with voicemail: my messages are good unless the
incoming call comes from isdn, which means via the avm fritz!card. In
this case (and in this case only) the message is disjointed and I can
hear at most 1 second out of a 1 minute message.
If the message comes from TDM400 then the message is perfect (even
though I still have a problem to detect the end of the call but that's
no big deal)
If the incoming call is answered (and not sent to voicemail because busy
or unavail) the sound is perfect.
I hope you'll be able to help me.
Thanks
Benjamin SEBBAH
ADUNEO France
Here are my config files:
</etc/asterisk/capi.conf>
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=fr ;set default language
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
isdnmode=DID ;'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
controller=1 ;capi controller number to use
group=9 ;dialout group
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
echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
echocancelold=yes;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
echotail=64 ;echo cancel tail setting
devices=2 ;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
and the interesting lines from </etc/asterisk/extensions.conf>:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
PIERRE=Zap/1
MARC=SIP/marc
PATRICK=Zap/3
PROSPECT=Zap/2
OPENSPACE=Zap/4
FT_FREE=Zap/5
FT_ALICE=Zap/6
VOIP_FREE=Zap/7
VOIP_ALICE=Zap/8
NUMERIS=CAPI/ISDN1
[macro-repondeur]
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},15,rWw) ; Ring the interface,
15 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
;exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
;exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to
start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else
as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user
into VoicemailMain
[capi-in]
;standard: fait tout sonner
exten => 3090,1,Answer;
;exten => 3090,2,Macro(repondeur,8427,${OPENSPACE}&${MARC}&${PIERRE});
exten => 3090,2,Macro(repondeur,8427,${OPENSPACE}&${PIERRE});
;Service technique
exten => 3091,1,Answer;
;exten => 3091,2,Macro(repondeur,3091,${OPENSPACE}&${MARC});
exten => 3091,2,Macro(repondeur,3091,${OPENSPACE});
;Service commercial
exten => 3092,1,Answer;
exten => 3092,2,Macro(repondeur,3092,${PATRICK});
;Direction technique
exten => 3093,1,Answer;
;exten => 3093,2,Macro(repondeur,3093,${MARC});
exten => 3093,2,Macro(repondeur,3093,${OPENSPACE});
;non assigne pour le moment fait sonner uniquement le DECT
exten => 3094,1,Answer;
exten => 3094,2,Macro(repondeur,3094,${OPENSPACE});
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
