[Asterisk-Users] Unable to create channem of type 'Zap'

[Wim Venneman]

Thanks for the help !

Made changes, still the same message.
I have two NIC's with IRQ 11
The FXO card has IRQ10 (and no other card has IRQ10)

Wim


----- Original Message ----- 
From: "Brent Franks" <mwless at mindworks.net>
To: <asterisk-users at lists.digium.com>
Sent: Monday, February 23, 2004 10:21 PM
Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'


> Wim, I made some changes to your Zapata.conf and zaptel.conf config
> files below.
> 
> Hope this helps.
> 
> Also, do a less /proc/interrupts and see if the card is on it's own IRQ.
> 
> - Brent
> 
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Wim Venneman
> Sent: Monday, February 23, 2004 3:10 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
> 
> Can anyone help me, (after a two day search, also on the mailing list)
> I have the following situation:
> Asterisk works fine, until I added a FXO card. (Digium)
> When I tried to call to the pstn I have the following error
> Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
> NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
> CHANNEL OF TYPE 'ZAP'
> == Everyone is busy at this time
> When I start Asterisk I have no error
> Only the following isn't right: 
> ZAP SHOW CHANNELS = No channels 
> modprobe wcfxo = ok (no errors)
> I have following config.
> ZAPATA
> [channels]
> language=en
> group=1
> pickupgroup=1
> context=incoming
> signalling=fxs_ks
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> immediate=yes
> musiconhold=default
> channel = 1
> 
> ZAPTEL 
> loadzone = us
> defaultzone = us
> fxsks = 1
> 
> EXTENSION
> [incoming]
> exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr)
> [outgoing]
> exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
> 
> IN [SIP]
> include => outgoing
> I'm don't know what I can change to the config.
> Anyone an idea
> Thanks,
> Wim
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
> 
>