[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});
___
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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
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});
___
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Re: [Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
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});
___
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Re: [Asterisk-Users] Asterisk voicemail problem with isdn avm fritz!card
On Mon, 19 Jun 2006, Benjamin Sebbah wrote:
- Original Message -
From: Armin Schindler [EMAIL PROTECTED]
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.
Ah, sure. I think it's just the echosquelch setting. echocancelold applies
for Eicon cards only and echosquelch causes frame-length changes.
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
[Asterisk-Users] Asterisk voicemail problem
Hi there, im having some troubles with my asterisk service, sometimes when im trying to make an outbound call, to any of the phones configured on the asterisk box,it enters inmediatly to voicemail and then hungs up. After that its necessary tostopthe service and putting up again manually.
Here is a piece of my log file when a call is trying to incoming:
"Jun 9 06:30:16 NOTICE[1125329728]: chan_sip.c:4879 handle_response: Peer '1366' is now REACHABLE!Jun 9 06:30:31 WARNING[1125329728]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request)^M -- Executing Goto("Zap/1-1", "4222760|s|1") in new stack^M -- Goto (4222760,s,1)^M -- Executing BackGround("Zap/1-1", "welcome-4222760") in new stack^M -- Accepting call from '16227735' to '4222760' on channel 1, span 1^M -- Playing 'welcome-4222760' (language 'en')^M -- Executing BackGround("Zap/1-1", "menu-4222760") in new stack^M -- Playing 'menu-4222760' (language 'en')Jun 9 06:30:42 WARNING[1125329728]: chan_sip.c:49
5
retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request)^M == CDR updated on Zap/1-1^M -- Executing Dial("Zap/1-1", "SIP/405|20|t") in new stack^M -- Called 405Jun 9 06:30:42 WARNING[1226204480]: channel.c:1858 ast_channel_make_compatible: No path to translate from SIP/405-db6d(256) to Zap/1-1(72)Jun 9 06:30:42 WARNING[1226204480]: chan_sip.c:1322 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 256/256)Jun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJ
un
9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voiceJun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voice^M == No one is available to answer at this time^M -- Executing VoiceMail2("Zap/1-1", "u4222760405") in new stack^M -- Playing 'vm-theperson'
(language 'en')^M -- Playing 'digits/4' (language 'en')^M -- Playing 'digits/2' (language 'en')^M -- Playing 'digits/2' (language 'en')^M -- Playing 'digits/2' (language 'en')^M -- Playing 'digits/7' (language 'en')^M -- Playing 'digits/6' (language 'en')^M -- Playing 'digits/0' (language 'en')^M -- Playing 'digits/4' (language 'en')^M -- Playing 'digits/0' (language 'en')^M -- Playing 'digits/5' (language 'en')^M -- Playing 'vm-isunavail' (language 'en')"
I dont know whatthe message "wait for answer: Unable to forward voice" does mean?. Every time that a call is trying to incoming appears the same log blockshown above.
I dont know what the problem is. Any help may be useful.
Thanks for your help.
Carlos Andres Medina
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