Simone Cittadini
2005-Aug-26 05:21 UTC
[Asterisk-Users] bridging sip to capi, no playtones back to caller
I've the following setup :
sip phone -> ser (auth and routing) -> asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear
anything till the called answer.
this is the output from a test call :
-- Executing Playtones("SIP/2.7.184.61-08152880",
"dial") in new stack
-- Executing Dial("SIP/2.7.184.61-08152880",
"CAPI/02myisdnnum:347callednum") in new stack
-- creating pipe for PLCI=-1
> sent CONNECT_REQ MN =0x193
-- Called 02myisdnnum:347callednum
-- CAPI[contr1/02myisdnnum]/2 is making progress passing it to
SIP/2.7.184.61-08152880
-- CAPI[contr1/02myisdnnum]/2 is ringing
> sent FACILITY_REQ (PLCI=0x101)
-- CAPI[contr1/02myisdnnum]/2 answered
== Spawn extension (default, 347callednum, 2) exited non-zero on
'SIP/2.7.184.61-08152880'
asterisk-pri-1:/etc/asterisk # cat extensions.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten => _X.,1,Playtones(ring)
exten => _X.,2,Dial,CAPI/0226265583:${EXTEN}
exten => _X.,3,HangupSIP/2.7.184.61-08152880
-- CAPI Hangingup
> sent DISCONNECT_B3_REQ NCCI=0x10101
> sent DISCONNECT_REQ PLCI=0x101
-- removed pipe for PLCI = 0x101
asterisk-pri-1:/etc/asterisk # cat sip.conf
[general]
context=default
port=5060
bindaddr=192.168.1.101
srvlookup=no
canreinvite=no
disallow=all
allow=alaw
asterisk-pri-1:/etc/asterisk # cat capi.conf
[general]
nationalprefix=0
internationalprefix=0039
rxgain=0.8
txgain=0.8
[interfaces]
msn=02myisdnnumber
incomingmsn=*
controller=1
softdtmf=0
context=default
callgroup=1
mode=immediate
devices=2
asterisk-pri-1:/etc/asterisk # cat indications.conf
[general]
country=it
[it]
description = Italy
ringcadence = 1000,4000
dial = 425/600,0/1000,425/200,0/200
busy = 425/500,0/500
ring = 425/1000,0/4000
congestion = 425/200,0/200
callwaiting = 425/200,0/600,425/200,0/10000
dialrecall = 470/400,425/400
record = 1400/400,0/15000
info =
!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0
Armin Schindler
2005-Aug-26 07:29 UTC
[Asterisk-Users] bridging sip to capi, no playtones back to caller
On Fri, 26 Aug 2005, Simone Cittadini wrote:> I've the following setup : > > sip phone -> ser (auth and routing) -> asterisk with capi isdn > > when I call a pstn number everything works fine, but I can't hear anything > till the called answer.If you want tones from isdn before the connection is established, you need to set 'early-B3'. With older chan_capi versions, you need to put 'b' or 'B' at the beginning of your 'callednum'. See README of chan_capi. If you want to use newer chan_capi, have a look at sourceforge.net. Armin> this is the output from a test call : > > -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack > -- Executing Dial("SIP/2.7.184.61-08152880", > "CAPI/02myisdnnum:347callednum") in new stack > -- creating pipe for PLCI=-1 > > sent CONNECT_REQ MN =0x193 > -- Called 02myisdnnum:347callednum > -- CAPI[contr1/02myisdnnum]/2 is making progress passing it to > SIP/2.7.184.61-08152880 > -- CAPI[contr1/02myisdnnum]/2 is ringing > > sent FACILITY_REQ (PLCI=0x101) > -- CAPI[contr1/02myisdnnum]/2 answered > == Spawn extension (default, 347callednum, 2) exited non-zero on > 'SIP/2.7.184.61-08152880' > > asterisk-pri-1:/etc/asterisk # cat extensions.conf > > [general] > static=yes > writeprotect=yes > [globals] > [default] > exten => _X.,1,Playtones(ring) > exten => _X.,2,Dial,CAPI/0226265583:${EXTEN} > exten => _X.,3,HangupSIP/2.7.184.61-08152880 > -- CAPI Hangingup > > sent DISCONNECT_B3_REQ NCCI=0x10101 > > sent DISCONNECT_REQ PLCI=0x101 > -- removed pipe for PLCI = 0x101 > > > asterisk-pri-1:/etc/asterisk # cat sip.conf > > [general] > context=default > port=5060 > bindaddr=192.168.1.101 > srvlookup=no > canreinvite=no > disallow=all > allow=alaw > > > asterisk-pri-1:/etc/asterisk # cat capi.conf > > [general] > nationalprefix=0 > internationalprefix=0039 > rxgain=0.8 > txgain=0.8 > [interfaces] > msn=02myisdnnumber > incomingmsn=* > controller=1 > softdtmf=0 > context=default > callgroup=1 > mode=immediate > devices=2 > > asterisk-pri-1:/etc/asterisk # cat indications.conf > > [general] > country=it > [it] > description = Italy > ringcadence = 1000,4000 > dial = 425/600,0/1000,425/200,0/200 > busy = 425/500,0/500 > ring = 425/1000,0/4000 > congestion = 425/200,0/200 > callwaiting = 425/200,0/600,425/200,0/10000 > dialrecall = 470/400,425/400 > record = 1400/400,0/15000 > info > !950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,!950/330,!1400/330,!1800/330,!0/1000,0 > > > > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >