Benjamin Sebbah
2006-Jun-19  01:42 UTC
[Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card
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});
Armin Schindler
2006-Jun-19  04:48 UTC
[Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:> 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 never heard of such a problem before. Can you please create a log of such a call with set verbose 9 capi debug (might be big) Armin> 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 >
Benjamin Sebbah
2006-Jun-19  06:07 UTC
[Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card
----- Original Message ----- From: Armin Schindler <armin@melware.de> Date: Monday, June 19, 2006 1:48 pm Subject: Re: [Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card> On Mon, 19 Jun 2006, Benjamin Sebbah wrote: > > 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 never heard of such a problem before. Can you please create a log > of such > a call with > set verbose 9 > capi debug > (might be big) > > Armin >Actually I have just found a solution: in capi.conf I've changed: rxgain=0.8 txgain=0.8 echosquelch=1 echocancelold=yes to rxgain=1 txgain=0.8 echosquelch=2 echocancelold=no and this works. Thanks for your help.> > 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 > > > _______________________________________________ > --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 >