asterisk users - Feb 2004 - 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
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
group=1
pickupgroup=1
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

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On Mon, 2004-02-23 at 17:10, Wim Venneman wrote:
> 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
> [channels]
> language=en
> 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
> group=1
> pickupgroup=1
> immediate=yes
> musiconhold=default channel => 1
                 ^^^^^^^

is this a typo? If not, the channel => 1 should go on a line of its own.

-- 
Nicolas Gudino <nicolas@house.com.ar>
House Internet S.R.L.

Make sure you run a ztcfg after you do a modprobe.

ztcfg will configure (or bring up) the zap channels on zaptel interface
cards.  Do this before starting * and after the modprobe.

(You may also do a ztcfg -v to see whats configured)

- Brent

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Wim Venneman
Sent: Monday, February 23, 2004 3:10 PM
To: asterisk-users@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
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
group=1
pickupgroup=1
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

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@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Wim Venneman
Sent: Monday, February 23, 2004 3:10 PM
To: asterisk-users@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

Wim,
	Made one more change below in Zapata.conf
 It should be channel => 1

-----Original Message-----
From: Wim Venneman [mailto:wim.venneman@skynet.be] 
Sent: Monday, February 23, 2004 4:46 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'

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@mindworks.net>
To: <asterisk-users@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@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Wim
Venneman
> Sent: Monday, February 23, 2004 3:10 PM
> To: asterisk-users@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
> 
> _______________________________________________
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I see that in queues.conf there is a maxlen variable to control the maximum 
size of the queue.  So...if you set the queue to a maxlen = 3...my test 
caller gets dead air if they are queued to a queue with 3 calls already in 
the queue.

I thought I could increment a variable each time a call is queued and 
decrement a variable when the call is connected to the agent...however I do 
not know how to build such a structure in extensions.conf to make this 
work.  It also *seems* like when an agent releases/hangs up/finishes a 
call...that the incoming caller is disconnected in such a way that 
additional steps in the dialing plan in extensions.conf are not processed.

Does anyone have a sample extensions.conf I can see that does something 
like this?

I thought I would try to give call center managers the ability to dial an 
extension, be authenticated, and then enter a number of their choice to 
allow them to set how many calls can be in a given queue so that if there 
are more agents available...the queue can take more calls...and when fewer 
agents are available...callers might hear a greeting indicating delays and 
be given the option to leave voice mail.    Does that sound like a 
reasonable idea?

I thought when I feel really crafty I would make a web interface in 
ColdFusion ( I do not speak PHP yet) and have Asterisk copy config files 
generated by my ColdFusion application from a cronjob to update the running 
Asterisk config.

Thanks in advance for any help...this list is great

Jeff