asterisk users - Jul 2004 - Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'

Hi,

I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box
(customized kernel version 2.4.24). I want calls from my SIP soft-phones
to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap
HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc.
I've read everything I've found at www.voip-info.org, then I've
downloaded
the latest bri-stuff.0.1.0-RC2g (released just today!) and started the
installation.

My extensions.conf is:

	[default]
	...cut...
	ignorepat => 9
	exten => 9,1,Dial(Zap/g1/)	; direct outbound dialing
	exten => 9,2,Congestion

Here's my zaptel.conf:

	loadzone=it
	defaultzone=it

	span=1,1,3,ccs,ami
	bchan=1-2
	dchan=3

Here's my zapata.conf:

	[channels]
	;
	; Default language
	;
	;language=en
	;
	; Default context
	;
	;
	switchtype = euroisdn
	; p2mp TE mode
	signalling = bri_cpe_ptmp

	pridialplan=local
	prilocaldialplan=local
	echocancel=yes
	immediate=yes
	group = 1
	context=default
	channel => 1-2

Here's my channels map:

	Zaptel Configuration
	=====================
	SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)

	Channel map:

	Channel 01: Individual Clear channel (Default) (Slaves: 01)
	Channel 02: Individual Clear channel (Default) (Slaves: 02)
	Channel 03: D-channel (Default) (Slaves: 03)

	3 channels configured.

Finally, here's my sip.conf:

	[general]
	context=default                 ; Default context for incoming calls
	...cut...
	disallow=all                    ; First disallow all codecs
	allow=gsm                       ; Allow codecs in order of preference
	allow=ulaw
	allow=alaw

	...cut...

	[alessandro]
	type=friend
	username=alessandro
	secret=bissoli
	host=dynamic
	dtmfmode=rfc2833
	disallow=all
	allow=gsm
	allow=alaw
	allow=ulaw



When I run asterisk and dial from the SIP phone I get this error:
   
    -- Executing Dial("SIP/alessandro-0f69", "Zap/g1/") in
new stack Jul
26 14:27:31 NOTICE[311313]: app_dial.c:711 dial_exec: Unable to create channel
of type 'Zap'
  == Everyone is busy/congested at this time
    -- Executing Congestion("SIP/alessandro-0f69", "") in
new stack
  == Spawn extension (default, 9, 2) exited non-zero on
'SIP/alessandro-0f69'

I really don't know what is wrong! Do you have any hints, please?

Alex

__________________________________________________________________
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Non perdere la promozione valida fino al 27 luglio. Per te gratis il modem
in comodato e l'attivazione. In piu' navighi a soli 1,5 euro l'ora
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primi tre mesi. Cosa aspetti? Attivala subito!
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I got things running with ISDN4Linux
See configuration example below, I found it exploring the WIKI-site.
(to make an outgoing call users need to press 0*[number to call])
Hope this is useful.

modem.conf:
-------------
[interfaces]
context=remote
noload => chan_modem.so
driver=i4l
language=nl
type=autodetect
dialtype=tone
mode=immediate

msn=[your MSN]
incomingmsn=[MSN1],[MSN2],...
device => /dev/ttyI0
device => /dev/ttyI1

extensions.conf
---------------
[remote]
include => local-sip

exten => s,1,Wait(20)
exten => s,2,Answer
exten => s,3,DigitTimeout(10)           ; Set Digit Timeout to 10 seconds
exten => s,4,ResponseTimeout(20)        ; Set Response Timeout to 20 seconds
exten => s,5,Background(vm-extension)   ; Ask them for the extension they
want

[dial-via-isdn]
exten => _0*XXX.,1,Dial(Modem/ttyI0:${EXTEN:2},30,r)
exten =>
_0*XXX.,2,Playback(asterisk-sounds/sounds/the-party-you-are-calling)
exten => _0*XXX.,3,Playback(asterisk-sounds/sounds/is-curntly-unavail)
exten => _0*XXX.,4,Wait(2.5)
exten => _0*XXX.,5,Playtones(congestion)
exten =>
_0*XXX.,102,Playback(asterisk-sounds/sounds/the-party-you-are-calling)
exten => _0*XXX.,103,Playback(asterisk-sounds/sounds/is-curntly-busy)
exten => _0*XXX.,104,Wait(2.5)
exten => _0*XXX.,105,Playtones(congestion)
exten => _0*XXX.,106,Hangup


<kleis-asterisk-dev@tiscali.it> schreef in bericht
news:4104C030000024DD@mail-3.tiscali.it...
Hi,

I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box
(customized kernel version 2.4.24). I want calls from my SIP soft-phones
to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap
HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc.
I've read everything I've found at www.voip-info.org, then I've
downloaded
the latest bri-stuff.0.1.0-RC2g (released just today!) and started the
installation.

Try 

	ignorepat => 9
	exten => _9.,1,Dial(Zap/g1/${EXTEN:1})	; direct outbound
dialing
	exten => _9.,2,Congestion

Best Regards 
Tim Robinson 
Tools Development Manager 
Motorola Ltd 
Midpoint 
Alencon Link 
BASINGSTOKE 
RG21 7PL 
United Kingdom 
Tel.   +44 1256 790472 
Fax    +44 1256 790190 
Mobile +44 7785 300316 



-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
kleis-asterisk-dev@tiscali.it
Sent: 26 July 2004 15:36
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based PCI
ISDN card): Unable to create channel of type 'Zap'


Hi,

I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box
(customized kernel version 2.4.24). I want calls from my SIP soft-phones
to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a
cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I
need zaphfc. I've read everything I've found at www.voip-info.org, then
I've downloaded the latest bri-stuff.0.1.0-RC2g (released just today!)
and started the installation.

My extensions.conf is:

	[default]
	...cut...
	ignorepat => 9
	exten => 9,1,Dial(Zap/g1/)	; direct outbound dialing
	exten => 9,2,Congestion

Here's my zaptel.conf:

	loadzone=it
	defaultzone=it

	span=1,1,3,ccs,ami
	bchan=1-2
	dchan=3

Hi

Il lun, 2004-07-26 alle 16:35, kleis-asterisk-dev@tiscali.it ha
scritto:
>     -- Executing Dial("SIP/alessandro-0f69", "Zap/g1/")
in new stack Jul
> 26 14:27:31 NOTICE[311313]: app_dial.c:711 dial_exec: Unable to create
channel
> of type 'Zap'
>   == Everyone is busy/congested at this time
>     -- Executing Congestion("SIP/alessandro-0f69", "")
in new stack
>   == Spawn extension (default, 9, 2) exited non-zero on
'SIP/alessandro-0f69'
> 
> I really don't know what is wrong! Do you have any hints, please?
just read what you see: unable to create channel of type Zap.
that seems that asterisk isn't loading chan_zap...
what says zap show channels on the cli?
or you see chan_zap.so in show modules ?

also, your extension.conf is wrong.
instead of
ignorepat => 9
exten => 9,1,Dial(Zap/g1/)      ; direct outbound dialing
exten => 9,2,Congestion

use
exten => _9X.,1,Dial(Zap/g1/${EXTEN:1})      ; direct outbound dialing
exten => _9X.,2,Congestion

then read the docs!

Matteo

-- 
****************************************
Matteo Brancaleoni
System Administrator
mbrancaleoni@espia.it
****************************************
EspiA Srl - e*solution provider
Via Pascoli, 37
20129 Milano - Italy
SIP:matteo@sip.voismart.it
Tel. +39 0270633354
Fax. +39 0245487890
IAXTEL: 17005662458
http://www.espia.it
****************************************

> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of Sjaakie Helderhorst
> Sent: Monday, July 26, 2004 5:14 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Re: Can't dial SIP<->EuroISDN (HFC-S
based
PCI
> ISDN card): Unable to create channel of type 'Zap'
> 
> I got things running with ISDN4Linux
> See configuration example below, I found it exploring the WIKI-site.
> (to make an outgoing call users need to press 0*[number to call])
> Hope this is useful.
It seems an interesting solution, but I need echo cancellation and so I
have to use zaphfc.

Thanks,

Alessandro

> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of kleis-asterisk-dev@tiscali.it
> Sent: Monday, July 26, 2004 4:36 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based
PCI
ISDN
> card): Unable to create channel of type 'Zap'
> 
> Hi,
> 
> I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux
box
> (customized kernel version 2.4.24). I want calls from my SIP
soft-phones
> to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a
> cheap
> HFC-S based PCI ISDN card connected to the NT1+ interface, so I need
> zaphfc.
> I've read everything I've found at www.voip-info.org, then I've
downloaded
> the latest bri-stuff.0.1.0-RC2g (released just today!) and started the
> installation.
I still have the problem! I really have no idea about what to do! Any
suggestion would be greatly appreciated.

Thanks,

Alex

> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of kleis-asterisk-dev@tiscali.it
> Sent: Monday, July 26, 2004 4:36 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based
PCI
ISDN
> card): Unable to create channel of type 'Zap'
Hi,

The following is from Asterisk's log (asterisk -vvvvgc | tee
asterisk.log):

[chan_skinny.so] => (Skinny Client Control Protocol (Skinny))
  == Parsing '/etc/asterisk/skinny.conf': Found
Jul 28 12:34:29 WARNING[16384]: chan_skinny.c:2584 reload_conf
ig: Unable to get our IP address, Skinny disabled
  == Registered channel type 'Skinny' (Skinny Client Control Protocol
(Skinny))
 [chan_oss.so] => (OSS Console Channel Driver)
  == Console is full duplex
  == Registered channel type 'Console' (OSS Console Channel Driver)
  == Parsing '/etc/asterisk/oss.conf': Found
Jul 28 12:34:29 WARNING[163850]: chan_oss.c:238 sound_thread
: Read error on sound device: Resource temporarily unavailable

Do you think that such warnigs may be somehow related to "Unable to
create channel of type 'Zap'"? (Soundcard is an onboard VIA chipset
based card)

Thanks,

Alex