Hello.
I have asterisk with an old avm b1 v3.0 configured and working with capi
channel . The isdn card is connected to an S0 isdn bus of a siemens
hipath 3000 version 4.0 It is possible to make outgoing calls and
receive too, but I think that there is some kind of signaling problem
when i call from an ip phone or soft ip phone because i do not get ring
or busy tone calling to any PBX phone, but if i get outside line (i.e.
calling a cell phone at PSTN) then i can hear the busy or ring tones.
So, if i call to any pbx phone i do not hear anything until someone
picks up the phone and if it is busy i do not know and after a while i
get the normal call clearing and the call is finished.
IP PHONE -----> B1 (ISDN) chan capi AT * ----------> S0 BUS HIPATH
---------> PSTN
Does anybody know any useful trick to solve this?
I know that may be it is a signaling problem and isdn related question,
but if someone can help me it would be great.
Pardon my bad English and thanks.
Some configured parameters:
****************************************************************************
;
;capi.conf
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=es
; interface sections ...
[ISDN1]
;ntmode=yes
isdnmode=msn
incomingmsn=620
controller=1
group=1 ;dialout group
;prefix=0
softdtmf=on ;enable/disable software dtmf detection, recommended
for AVM cards
accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in ;context for incoming calls
holdtype=hold ;
;immediate=yes
;echosquelch=1
;echocancel=yes
;echotail=64
bridge=yes
callgroup=1
language=es
***********************************************************************
;
;indications.conf
;
[general]
country=es
[es]
description = Spain
ringcadence =1500,3000
dial = 423
busy =425/200,0/200
congestion = 425/200,0/200,425/200,0/200,425/200,0/600
callwaiting = 425/175,0/175,425/175,0/3500
dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425
record = 1400/500,0/15000
info = 950/330,0/1000
dialout = 500
*********************************************************************
;
; part of dial extensions.conf for dialing outisde
;RDSI=620 is the number for the isdn S0 bus
;
[capi-out]
exten => _0.,1,NoOp("Salida a la calle" ${CALLERID})
exten => _0.,2,Dial(CAPI/g1/${RDSI},20)
...
**********************************************************************
Thanks.
Ricardo
Armin Schindler
2006-Apr-20 02:38 UTC
[Asterisk-Users] avm b1with chan capi and siemens hipath
Did you try the /b option of Dial() with capi? This enables early-b3, whcih gives you progress tones from the ISDN line. Armin On Thu, 20 Apr 2006, Ricardo wrote:> Hello. > I have asterisk with an old avm b1 v3.0 configured and working with capi > channel . The isdn card is connected to an S0 isdn bus of a siemens hipath > 3000 version 4.0 It is possible to make outgoing calls and receive too, but I > think that there is some kind of signaling problem when i call from an ip > phone or soft ip phone because i do not get ring or busy tone calling to any > PBX phone, but if i get outside line (i.e. calling a cell phone at PSTN) then > i can hear the busy or ring tones. > > So, if i call to any pbx phone i do not hear anything until someone picks up > the phone and if it is busy i do not know and after a while i get the normal > call clearing and the call is finished. > > IP PHONE -----> B1 (ISDN) chan capi AT * ----------> S0 BUS HIPATH ---------> > PSTN > > Does anybody know any useful trick to solve this? > I know that may be it is a signaling problem and isdn related question, but if > someone can help me it would be great. > > Pardon my bad English and thanks. > > Some configured parameters: > **************************************************************************** > ; > ; capi.conf > ; > > ; general section > > [general] > nationalprefix=0 > internationalprefix=00 > rxgain=0.8 > txgain=0.8 > language=es > ; interface sections ... > > [ISDN1] ;ntmode=yes isdnmode=msn incomingmsn=620 controller=1 > group=1 ;dialout group > ;prefix=0 softdtmf=on ;enable/disable software dtmf detection, > recommended for AVM cards > accountcode= ;Asterisk accountcode to use in CDRs > context=capi-in ;context for incoming calls > holdtype=hold ; > ; immediate=yes echosquelch=1 echocancel=yes echotail=64 > bridge=yes callgroup=1 language=es > *********************************************************************** > ; > ; indications.conf > ; > [general] > country=es > > [es] > description = Spain > ringcadence =1500,3000 > dial = 423 > busy =425/200,0/200 > congestion = 425/200,0/200,425/200,0/200,425/200,0/600 > callwaiting = 425/175,0/175,425/175,0/3500 > dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 > record = 1400/500,0/15000 > info = 950/330,0/1000 > dialout = 500 > > ********************************************************************* > ; > ; part of dial extensions.conf for dialing outisde > ; RDSI=620 is the number for the isdn S0 bus > ; > [capi-out] > exten => _0.,1,NoOp("Salida a la calle" ${CALLERID}) > exten => _0.,2,Dial(CAPI/g1/${RDSI},20) > ... > > ********************************************************************** > Thanks. > Ricardo > _______________________________________________ > --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 >
2006/4/20, Armin Schindler <armin@melware.de>:> > Did you try the /b option of Dial() with capi? > This enables early-b3, whcih gives you progress tones from the ISDN line. > > Armin > > > That was the reason!I though that i tested that option before but may be i made some mistake. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060420/6e121c27/attachment.htm