[Asterisk-Users] Re: Problems with incoming calls on a E1 ISDN PRI

[Roberto Reiner Uhry]

Hi,

I have an Asterisk installed on a FC3 with a Digium e100p card and an
E1 (ISDN PRI).
I'm in Brazil and using Embratel as carrier.

After few troubles I get it working to make calls, from a SIP channel
to an Fone through the carrier.  But when I receive a call, this one
is transfered to the SIP channel but when I answere this one stay
quiet.

Does anybody have any ideai about how can I solve this problem?
 
/etc/zaptel.conf
loadzone = us
defaultzone = us
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
alaw=1-31 --- I also tried with/without this line

/etc/asterisk/extensions.conf
[from-sip-external]
exten => _.,1,AbsoluteTimeout(15)
exten => _.,2,Congestion
exten => _.,3,Hangup

[from-sip]
exten => _66XX,1,Dial(SIP/${EXTEN},20,t)
exten => _66XX,2,Hangup()
exten => _0.,1,Dial(Zap/g1/${EXTEN:1})

[default]
exten => 6689,1,Dial(SIP/6689,20,t)
exten => 6689,2,Hangup()

/etc/asterisk/sip.conf
[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=alaw
allow=gsm
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown

;include sip_nat.conf
;include sip_additional.conf

[6689]
context = from-sip
username = 6689
secret = 6689
host = dynamic
type = friend
regexten = 6689
allow=ulaw

/etc/asterisk/zapata.conf
[trunkgroups]
trunkgroup => 1,16
spanmap => 1,1,1

[channels]
switchtype=euroisdn
signalling=pri_cpe
language=us
defaultzone=us
group = 1
musiconhold = default
echocancel=yes
channel => 1-15,17-31