[Asterisk-Users] Re: zap channel won't send/receive calls

[fahmy kadiri]

Hi,
I have a fxo card configured in my asterisk pbx.. but cant seem to
make calls from POTS ---> sip
or from sip ----> pots

when I place a call, asterisk seems to begin the call processing

-- Executing Dial("SIP/fahmy-452d", "Zap/1/4168880686") in new stack
   -- Called 1/4168880686
   -- Zap/1-1 answered SIP/fahmy-452d

and thats where it stops... no dial tone, the other side is not
ringing, and nothing is happening on the line
finally when I hang up this is what I see on console

 -- Hungup 'Zap/1-1'
 == Spawn extension (from-sip-internal, 94168880686, 1) exited
non-zero on 'SIP/fahmy-452d'
   -- Executing Hangup("SIP/fahmy-452d", "") in new stack
 == Spawn extension (from-sip-internal, h, 1) exited non-zero on
'SIP/fahmy-452d'

and if I make a call into my asterisk box from an external number, the
call is detected, and a few seconds later it is hung up... still no
dial tone

-- Starting simple switch on 'Zap/1-1'
   -- Executing PrivacyManager("Zap/1-1", "") in new stack
   -- CallerID Present: Skipping
   -- Executing Dial("Zap/1-1", "SIP/1100&SIP/1400|30") in new stack
Nov 14 20:32:45 WARNING[1632]: chan_sip.c:1386 create_addr: No such host: 1100
Nov 14 20:32:45 NOTICE[1632]: app_dial.c:743 dial_exec: Unable to
create channel of type 'SIP'
Nov 14 20:32:45 WARNING[1632]: chan_sip.c:1386 create_addr: No such host: 1400
Nov 14 20:32:45 NOTICE[1632]: app_dial.c:743 dial_exec: Unable to
create channel of type 'SIP'
 == Everyone is busy/congested at this time
   -- Executing VoiceMail2("Zap/1-1", "b1100") in new stack
   -- Playing 'vm-theperson' (language 'en')
-- Recording the message
   -- x=0, open writing:
/var/spool/asterisk/voicemail/local/1100/INBOX/msg0000 format: wav,
0x8107580

and even though voicemail is being played i can't seem to hear it,
then the call gets disconnected before I have a chance to leave
voicemail

I've tried everything and this is my last resort... any help is
greatly appreciated

here is my dmesg output... sorry for the long email

dmesg output:
------------------
Zapata Telephony Interface Registered on major 196
PCI: Found IRQ 12 for device 00:09.0
PCI: Sharing IRQ 12 with 00:10.1
wcfxo: DAA mode is 'FCC'
Found a Wildcard FXO: Generic Clone
ISDN subsystem Rev: 1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1 loaded
HiSax: Linux Driver for passive ISDN cards
HiSax: Version 3.5 (module)
HiSax: Layer1 Revision 1.1.4.1
HiSax: Layer2 Revision 1.1.4.1
HiSax: TeiMgr Revision 1.1.4.1
HiSax: Layer3 Revision 1.1.4.1
HiSax: LinkLayer Revision 1.1.4.1
HiSax: Approval certification failed because of
HiSax: unauthorized source code changes

/etc/zaptel.conf
--------------------
fxsks = 1
loadzone = us
defaultzone = us

/etc/asterisk/zapata.conf
-----------------------
[channels]
language=en
context=from-analog
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
channel => 1