Hi all,
I want my voicemail to answer after 20 seconds, however there is a hangup after ~10 seconds. The telephones are still ringing, but for asterisk it looks like it got hung up.
I’m using a FRITZ!Card PCI withCAPI Driver Version 03.11.07 on debian/sid, kernel 2.6.17.11.
Here’s the capi debug when the hangup occurs:
INFO_IND ID=001 #0x5f6f LEN=0033
Controller/PLCI/NCCI = 0x1
InfoNumber = 0x1c
InfoElement = <91 a1 0f 02 02 d7 a3 06 06 04 00 82>g<01 0a 02 01 02>
INFO_RESP ID=001 #0x5f6f LEN=0012
Controller/PLCI/NCCI = 0x1
CAPI: INFO_IND no interface for PLCI=0x1
> CAPI: Command=INFO_IND,0x8492: no interface for PLCI=0x1, MSGNUM=0x5f6f!
Any idea?
Am I really the only one with that problem?
Please realevent part of extensions.conf
[et-voice-nas]
exten => nas,1,Wait(20)
exten => nas,2,Answer()
exten => nas,3,VoiceMail(nas@default)
exten => a,1,VoiceMailMain(nas@default)
However, it hangs up within priority 1.
Here is my capi.conf:
[code];
; 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’.
;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
msn=12345678
;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)
[/code]
Try this:
Exten => 123,1,Answer
exten => 123,n,Ringing
exten => 123,n,Wait(20)
exten => 123,n,Voicemail(box@context)
[quote=“Dovid”]Try this:
Exten => 123,1,Answer
[/quote]
Thx for the “Answer” 
However, I wouldn’t want the caller to be charged, if s/he hangsup before the 20s.
post a debug log fragment for a call. is Asterisk hanging up, or detecting a remote hangup ?
i’m lost wondering why you want someone to have to wait for 20 seconds of ringing before you send them to VM though !!
[code] INFO_IND ID=001 #0x5f6f LEN=0033
Controller/PLCI/NCCI = 0x1
InfoNumber = 0x1c
InfoElement = <91 a1 0f 02 02 d7 a3 06 06 04 00 82>g<01 0a 02 01 02>
INFO_RESP ID=001 #0x5f6f LEN=0012
Controller/PLCI/NCCI = 0x1
CAPI: INFO_IND no interface for PLCI=0x1
> CAPI: Command=INFO_IND,0x8492: no interface for PLCI=0x1, MSGNUM=0x5f6f!
[/code]
To me it looks like asterisk detects a hangup.
If I can’t answer the phone myself I want the answering machine to answer the call.
that’s an Asterisk /var/log/asterisk/full log file fragment ???
your extensions.conf context doesn’t indicate you’re doing a Dial to a device … there’s something you’re not telling about your setup.
No, it’s CAPI DEBUG. I’ll test it again and post the full log.
I was calling myself from the same ISDN.
I’ll use a different phone next time.