I've just installed an X100P, built the kernel module, and tried to use it
to make an outgoing call (via a phone connected to an ATA-186). However, I
just get a reorder tone and see this on the console:
-- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in
new stack
NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to
create channel of type 'Zap'
== Everyone is busy at this time
-- Executing Congestion("SIP/ata1-4409", "") in new
stack
My zapata.conf file is unchanged from the default, which seemed to be
appropriate for the X100P in this usage.
It's not clear to me why in particular it is unable to create the channel.
Is it a hardware problem? Software? Lunar phase?
[root@kotter]# lsmod | grep wcfxo
wcfxo 8352 0 (unused)
zaptel 178880 2 [wcfxo]
Is this what I should see?
miguel
/etc/zapata.conf
fxsks=1
loadzone=us
defaultzone=us
now reload your wcfxo module
When you run zttool, what do you see?
If your telephone line form the PSTN is plugged in it should read OK not RED
Please respond with the line from your extensions.conf to see if you are
dialing out of your Zap device properly
For example:
If you're dialing 9 to get out it would look something like this ( in this
instance I have used my zaptel for outbound calls as group 1)
/etc/asterisk/zapata.conf (should look something like this)
[channels]
group=1
stripmsd=0
signalling=fxs_ks
channel => 1
/etc/asterisk/extensions.conf (should look something like this)
[outbound]
exten => _9NXXXXXX,1,Dial,Zap/g1/${EXTEN:1}
Hope this helps
-Greg Merriweather
----- Original Message -----
From: "Miguel Cruz" <mnc-asterisk@u.nu>
To: <asterisk-users@lists.digium.com>
Sent: Friday, May 23, 2003 9:13 PM
Subject: [Asterisk-Users] Unable to create channel of type 'Zap'
> I've just installed an X100P, built the kernel module, and tried to use
it
> to make an outgoing call (via a phone connected to an ATA-186). However, I
> just get a reorder tone and see this on the console:
>
> -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212")
in new stack
> NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to
> create channel of type 'Zap'
> == Everyone is busy at this time
> -- Executing Congestion("SIP/ata1-4409", "") in new
stack
>
> My zapata.conf file is unchanged from the default, which seemed to be
> appropriate for the X100P in this usage.
>
> It's not clear to me why in particular it is unable to create the
channel.
> Is it a hardware problem? Software? Lunar phase?
>
> [root@kotter]# lsmod | grep wcfxo
> wcfxo 8352 0 (unused)
> zaptel 178880 2 [wcfxo]
>
> Is this what I should see?
>
> miguel
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Eduardo Goncalves
2003-May-29 06:57 UTC
[Asterisk-Users] Unable to create channel of type 'Zap'
Hi list,
I have the follow configuration:
==============extension.conf:
==============[pstn]
ignorepat => 0
exten => _0XXXXXXXX,1,Dial(${TRUNK}/${EXTEN:1})
[default]
exten => 120,1,Dial(IAX/eduardo@10.0.11.103)
include => pstn
But, when I dial from my gnophone something like 097991269, asterisk console
returns the fallow message:
NOTICE[245775]: File app_dial.c, Line 481 (dial_exec): Unable to create channel
of type 'Zap'
Could anyone help me?
thanks in advance
Eduardo
Eduardo Goncalves
2003-May-29 12:31 UTC
[Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 12:08:32 -0700 Andrew Gillham <gillham@vaultron.com> wrote:> Does it work without the group? e.g. Zap/1 > Also, does 'zap show channel 1' look ok? > > -Andrewyeap, I tried Zap/1 and it didn't work. :~( *CLI> zap show channel 1 Channel: 1 File Descriptor: 17 Span: 1 Extension: Context: default Caller ID string: Destroy: 0 Signalling Type: E & M Immediate Owner: <None> Real: <None> Callwait: <None> Threeway: <None> Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x0000 Actual Confmute: No *CLI> thanks Eduardo
Tim Davidson
2005-Jan-08 18:12 UTC
[Asterisk-Users] Unable to create channel of type 'Zap'
I've looked around so hard for days now for a solution to my problem.
I'm new to asterisk but I've managed to get an IAX coinnection working
to voiptalk.org, ive got sip to sipgate. But I just can't get my X100P
clone working.
I'm running Redhat 9
It's got IRQ 5 all to itself
ztcfg likes it :Zaptel Configuration
=====================
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
zapata.conf is as described in many howto's:
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1-4
immediate=no
busydetect=yes
busycount=7
; PSTN Home Line Context
context=telewest-incoming
signalling=fxs_ks
callerid=asreceived
channel=1
Extensions.conf has the following :
[telewest-incoming]
exten => s,1,Wait,2
exten => s,2(Dial,SIP/grandstream1,20,tr)
exten => s,3,Hangup
[telewest]
exten => _6X.,1,Dial,Zap/1/${EXTEN:1}
When dialing the speaking clok in the UK (123) with the prefix of 6 I get this
on the * console:
-- Executing Dial("SIP/grandstream1-75e0", "Zap/1/123") in
new stack
Jan 9 01:00:20 NOTICE[10689]: app_dial.c:746 dial_exec: Unable to create
channel of type 'Zap'
== Everyone is busy/congested at this time
Any Help would be verg greatfully received.
Regards
Tim Davidson
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050108/60793360/attachment.htm
VenkataRao Chimata
2005-Jan-15 10:29 UTC
[Asterisk-Users] Unable to create channel of type 'Zap'
Hi friends
I recently bought an X100P card and fixed it to my
PC.
I connected one port to an analog phone and the
other(the port which is supposed to be connected to
the telephone network) is left unconnected to anyone.
When tried to make a call from asterisk command prompt
to the phone I am getting the following error.
-- Executing Dial("ALSA/default", "Zap/1|20") in new
stack
Jan 15 18:28:45 NOTICE[1152795712]: app_dial.c:714
dial_exec: Unable to create channel of type 'Zap'
== Everyone is busy/congested at this time
When I typed zttool the card X100P is not listed.
Instead X101P is listed. Why I am getting this with an
X100p card.(I dont have X101P card)
I have the following in my configuration files.
zaptel.conf
fxsls=1
loadzone = us
defaultzone=us
zapatal.conf
[channels]
;
;
;
busydetect=yes
;
busycount=7
;
;
;
relaxdtmf=yes
;
callwaiting=yes
;
callwaitingcallerid=yes
;
threewaycalling=yes
;
transfer=yes
;
cancallforward=yes
;
;
;
usecallerid=yes
;
;
;
echocancel=yes
;
echocancelwhenbridged=yes
;
;
;
rxgain=0.0
;
txgain=0.0
;
;
;
group=1
;
pickupgroup=1
;
;
;
immediate=no
;
context=default
;
;
;
signalling=fxs_ls
;
callerid=asreceived
;
channel => 1
extensions.conf
[default]
exten => 5454,1,Dial(Zap/1,20)
Some more information
--------------------------
1. From the linux command line, I typed lspci and
got some message like "Tiger Jet"
2. the device is assigned an unique IRQ by the
proecessor.
3. wcfxo and zaptel are loaded properly.
4. chan_zap.so is loaded well when asterisk program is
executed.
Please help me.
I am trying with this for the last few days.
But I couldn't get any solution.
Please help me.
Thanks in advance
Regards
Venkat
__________________________________
Do you Yahoo!?
Yahoo! Mail - Find what you need with new enhanced search.
http://info.mail.yahoo.com/mail_250
VenkataRao Chimata
2005-Jan-15 23:03 UTC
[Asterisk-Users] Unable to create channel of type 'Zap'
Date: Sat, 15 Jan 2005 13:19:02 -0500 From: "Brent Franks" <mwless@mindworks.net> Subject: RE: [Asterisk-Users] Unable to create channel of type 'Zap' To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <200501151809.NAA11200@www.mindworks.net> Content-Type: text/plain; charset="US-ASCII"> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com[mailto:asterisk-users-> bounces@lists.digium.com] On Behalf Of VenkataRaoChimata> Sent: Saturday, January 15, 2005 12:30 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Unable to create channelof type 'Zap'> > Hi friends > > I recently bought an X100P card and fixed it to my > PC. > I connected one port to an analog phone and the > other(the port which is supposed to be connected to > the telephone network) is left unconnected toanyone.> When tried to make a call from asterisk commandprompt> to the phone I am getting the following error. > > > -- Executing Dial("ALSA/default", "Zap/1|20") innew> stack > Jan 15 18:28:45 NOTICE[1152795712]: app_dial.c:714 > dial_exec: Unable to create channel of type 'Zap' > == Everyone is busy/congested at this timeAfter you do modprobe wcfxo Are you running ztcfg -vvvv ? - Brent Yes I did ztcfg -vvvv also. And I got the following output Zaptel Configuration ===================== Channel map: Channel 01: FXS Loopstart (Default) (Slaves: 01) 1 channels configured. __________________________________ Do you Yahoo!? The all-new My Yahoo! - What will yours do? http://my.yahoo.com
Jaime Blanco
2005-Apr-20 11:33 UTC
[Asterisk-Users] Unable to create channel of type 'Zap'
Hi,
I just installed the asterisk and the X100P card. I can receive calls from
PSTN and it can ring on a Grandstream SIP Phone. From the SIP Phone I can
dial the demo extension on asterisk pbx. The issue is as soon as I try to
dial out 92714756 or another number I received the following message:
*CLI> -- Executing Dial("SIP/1001-2b93",
"Zap/g2/2714756") in new stack
Apr 20 02:27:40 NOTICE[245776]: app_dial.c:536 dial_exec: Unable to create
channel of type 'Zap'
== Everyone is busy at this time
-- Executing Congestion("SIP/1001-2b93", "") in new
stack
== Spawn extension (from-sip, 92714756, 2) exited non-zero on
'SIP/1001-2b93'
Zapata.conf is:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
immediate=no
context=default
signalling=fxs_ks
channel=1
extensions.conf
;
; Static extension configuration file, used by ; the pbx_config module. This
is where you configure all your ; inbound and outbound calls in Asterisk.
;
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite ;
this file when extensions are modified. Remember that all comments ; made
in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by ; CLI command
'save dialplan' too ; writeprotect=no
; You can include other config files, use the #include command (without the
';')
; Note that this is different from the "include" command that includes
contexts within ; other contexts. The #include command works in all asterisk
configuration files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be
referenced ;
in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid ; [globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:pass@provider
;
; Any category other than "General" and "Globals" represent
; extension
contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations ; thereof. If an
extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a ; literal. In
patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXXXXXX would match normal 7 digit dialings, ;
while _1NXXNXXXXXX would represent an area code plus phone number ;
preceeded by a one.
;
; Contexts contain several lines, one for each step of each ; extension,
which can take one of two forms as listed below, ; with the first form being
preferred. One may include another ; context in the current one as well,
optionally with a ; date and time. Included contexts are included in the
order ; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
; <time range>|<days of week>|<days of month>|<months>
;
;include => daytime|9:00-17:00|mon-fri|*|* ; ; ignorepat can be used to
instruct drivers to not cancel dialtone upon ; receipt of a particular
pattern. The most commonly used example is ; of course '9' like this:
;
;ignorepat => 9
; so that dialtone remains even after dialing a 9.
;
;
; Here are the entries you need to participate in the IAXTEL ; call routing
system. Most IAXTEL numbers begin with 1-700, but ; there are exceptions.
For more information, and to sign ; up, please go to www.gnophone.com or
www.iaxtel.com ; [iaxtel700] exten =>
_91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext
[trunkint]
;
; International long distance through trunk ; exten =>
_9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk ; exten =>
_91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface ; exten =>
_9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface ; exten =>
_91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion
[international]
;
; Master context for international long distance ; ignorepat => 9 include
=>
longdistance include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only ; ignorepat =>
9 include => default include => parkedcalls include => trunklocal
include =>
iaxtel700 include => trunktollfree include => iaxprovider ; ; You can use
an
alternative switch type as well, to resolve ; extensions that are not known
here, for example with remote ; IAX switching you transparently get access
to the remote ; Asterisk PBX ; ; switch =>
IAX2/user:password@bigserver/local
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring the
interface, 20 seconds maximum
exten => s,2,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten => s,3,Goto(default,s,1) ; If they
press #, return to start
exten => s,102,Voicemail(b${ARG1}) ; If busy,
send to voicemail w/ busy announce
exten => s,103,Goto(default,s,1) ; If they
press #, return to start
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct) ; Play some instructions
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,2,Goto(s,6)
exten => 3,1,SetLanguage(fr) ; Set language to french
exten => 3,2,Goto(s,5) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has ;
voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold
while..."
; (but skip if channel is not up)
exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,2,Voicemail(u1234) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the
demo"
exten => #,2,Hangup ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try
again"
;
; Create an extension, 500, for dialing the ; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the
Asterisk demo
exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo
site
exten => 500,4,Goto(s,6) ; Return to the start over message.
;
; Create an extension, 600, for evaulating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,2,Echo ; Do the echo test
exten => 600,3,Playback(demo-echodone) ; Let them know it's over
exten => 600,4,Goto(s,6) ; Start over
;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example) ; ;exten =>
1265,1,Dial(Phone/phone0,15) ;exten => 1265,2,Goto(s,5)
;[mainmenu]
; Example "main menu" context with submenu ; ;exten => s,1,Answer
;exten => s,2,Background(thanks) ; "Thanks for calling
press
1 for sales, 2 for support, ..."
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them
comfortable with 2 seconds of ringback
;exten => s,2,Wait,2
;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales
department. Press 1 for steve, 2 for..."
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
[default]
;
; By default we include the demo. In a production system, you ; probably
don't want to have the demo there.
;
include => demo
; Real extensions would go here. Generally you want real extensions to be 4
or 5 ; digits long (although there is no such requirement) and start with a
single ; digit that is fairly large (like 6 or 7) so that you have plenty of
room to ; overlap extensions and menu options without conflict. You can
alias them with ; names, too and use global variables
exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming
${MARK} is something like Zap/2
exten => mark,1,Goto(6275|1) ;
alias mark to 6275
exten => 6236,1,Macro(stdexten,6236,${WIL}) ; Ditto for
wil
exten => wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via ;
voicemailmain ; ;exten => 8500,1,VoicemailMain ;exten => 8500,2,Hangup ; ;
Or a conference room (you'll need to edit meetme.conf to enable this room) ;
;exten => 8600,1,Meetme,1234 ; ; Or playing an announcement to the called
party, as soon it answers ; ;exten =
8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "show applications"
at
your ; friendly Asterisk CLI prompt.
;
; 'show application <command>' will show details of how you ; use
that
particular application in this file, the dial plan.
;
[from-sip]
exten => 1001,1,Dial(SIP/1001)
exten => 1001,2,Wait(1)
exten => 1001,3,Answer
exten => 1001,4,Hangup
include => demo
include => local
;exten => _9X,1,Dial,Zap/1/${EXTEN:1}
;exten => _9X,2,Goto(102)
;exten => _9X,102,Congestion
;exten => _9X,103,Hangup
ZAPTEL.CONF
loadzone = us
#default = us
fxsks=1
Please, notice that "default=us" is commented since when I run ztcfg
-vvv it
gave me the following error:
root@knoppix:/etc# ztcfg -vvvv
Notice: Configuration file is /etc/zaptel.conf line 2: Unknown keyword
'default'
1 error(s) detected
I commented the line 2 and run ztcfg again and it worked without errors.
May it has no relationship with the error describe above.
Thanks.
Jaime
Jaime Blanco
President
Ximark Technologies, Inc.
Solutions for Keeping your Business Up
Phone: +507 271 5951 (Panama)
+1 928 752 1325 (USA)
Cell: +507 676 0623
Corporate email: <mailto:jaime.blanco@ximark.com> jaime.blanco@ximark.com
Personal email: <mailto:jaime@blanco.com> jaime@blanco.com
MSN ID: <mailto:blancolandau@hotmail.com>
blancolandau@hotmail.com
Jaime Blanco
2005-Apr-20 12:09 UTC
[Asterisk-Users] Unable to create channel of type 'Zap'
Hi,
I just installed the asterisk and the X100P card. I can receive calls from
PSTN and it can ring on a Grandstream SIP Phone. From the SIP Phone I can
dial the demo extension on asterisk pbx. The issue is as soon as I try to
dial out 92714756 or another number I received the following message:
*CLI> -- Executing Dial("SIP/1001-2b93",
"Zap/g2/2714756") in new stack
Apr 20 02:27:40 NOTICE[245776]: app_dial.c:536 dial_exec: Unable to create
channel of type 'Zap'
== Everyone is busy at this time
-- Executing Congestion("SIP/1001-2b93", "") in new
stack
== Spawn extension (from-sip, 92714756, 2) exited non-zero on
'SIP/1001-2b93'
Zapata.conf is:
[channels]
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
immediate=no
context=default
signalling=fxs_ks
channel=1
extensions.conf
;
; Static extension configuration file, used by ; the pbx_config module. This
is where you configure all your ; inbound and outbound calls in Asterisk.
;
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite ;
this file when extensions are modified. Remember that all comments ; made
in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by ; CLI command
'save dialplan' too ; writeprotect=no
; You can include other config files, use the #include command (without the
';')
; Note that this is different from the "include" command that includes
contexts within ; other contexts. The #include command works in all asterisk
configuration files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be
referenced ;
in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid ; [globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:pass@provider
;
; Any category other than "General" and "Globals" represent
; extension
contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations ; thereof. If an
extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a ; literal. In
patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
;
; For example the extension _NXXXXXX would match normal 7 digit dialings, ;
while _1NXXNXXXXXX would represent an area code plus phone number ;
preceeded by a one.
;
; Contexts contain several lines, one for each step of each ; extension,
which can take one of two forms as listed below, ; with the first form being
preferred. One may include another ; context in the current one as well,
optionally with a ; date and time. Included contexts are included in the
order ; they are listed.
;
;[context]
;exten => someexten,priority,application(arg1,arg2,...)
;exten => someexten,priority,application,arg1|arg2...
;
; Timing list for includes is
;
; <time range>|<days of week>|<days of month>|<months>
;
;include => daytime|9:00-17:00|mon-fri|*|* ; ; ignorepat can be used to
instruct drivers to not cancel dialtone upon ; receipt of a particular
pattern. The most commonly used example is ; of course '9' like this:
;
;ignorepat => 9
; so that dialtone remains even after dialing a 9.
;
;
; Here are the entries you need to participate in the IAXTEL ; call routing
system. Most IAXTEL numbers begin with 1-700, but ; there are exceptions.
For more information, and to sign ; up, please go to www.gnophone.com or
www.iaxtel.com ; [iaxtel700] exten =>
_91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext
[trunkint]
;
; International long distance through trunk ; exten =>
_9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk ; exten =>
_91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface ; exten =>
_9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface ; exten =>
_91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion
[international]
;
; Master context for international long distance ; ignorepat => 9 include
=>
longdistance include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only ; ignorepat =>
9 include => default include => parkedcalls include => trunklocal
include =>
iaxtel700 include => trunktollfree include => iaxprovider ; ; You can use
an
alternative switch type as well, to resolve ; extensions that are not known
here, for example with remote ; IAX switching you transparently get access
to the remote ; Asterisk PBX ; ; switch =>
IAX2/user:password@bigserver/local
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring the
interface, 20 seconds maximum
exten => s,2,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten => s,3,Goto(default,s,1) ; If they
press #, return to start
exten => s,102,Voicemail(b${ARG1}) ; If busy,
send to voicemail w/ busy announce
exten => s,103,Goto(default,s,1) ; If they
press #, return to start
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct) ; Play some instructions
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,2,Goto(s,6)
exten => 3,1,SetLanguage(fr) ; Set language to french
exten => 3,2,Goto(s,5) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has ;
voicemail, etc.
;
exten => 1234,1,Playback(transfer,skip) ; "Please hold
while..."
; (but skip if channel is not up)
exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,2,Voicemail(u1234) ; Unless busy
;
; # for when they're done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the
demo"
exten => #,2,Hangup ; Hang them up.
;
; A timeout and "invalid extension rule"
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try
again"
;
; Create an extension, 500, for dialing the ; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,2,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the
Asterisk demo
exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo
site
exten => 500,4,Goto(s,6) ; Return to the start over message.
;
; Create an extension, 600, for evaulating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,2,Echo ; Do the echo test
exten => 600,3,Playback(demo-echodone) ; Let them know it's over
exten => 600,4,Goto(s,6) ; Start over
;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
;
; Here's what a phone entry would look like (IXJ for example) ; ;exten =>
1265,1,Dial(Phone/phone0,15) ;exten => 1265,2,Goto(s,5)
;[mainmenu]
; Example "main menu" context with submenu ; ;exten => s,1,Answer
;exten => s,2,Background(thanks) ; "Thanks for calling
press
1 for sales, 2 for support, ..."
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them
comfortable with 2 seconds of ringback
;exten => s,2,Wait,2
;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales
department. Press 1 for steve, 2 for..."
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)
[default]
;
; By default we include the demo. In a production system, you ; probably
don't want to have the demo there.
;
include => demo
; Real extensions would go here. Generally you want real extensions to be 4
or 5 ; digits long (although there is no such requirement) and start with a
single ; digit that is fairly large (like 6 or 7) so that you have plenty of
room to ; overlap extensions and menu options without conflict. You can
alias them with ; names, too and use global variables
exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming
${MARK} is something like Zap/2
exten => mark,1,Goto(6275|1) ;
alias mark to 6275
exten => 6236,1,Macro(stdexten,6236,${WIL}) ; Ditto for
wil
exten => wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via ;
voicemailmain ; ;exten => 8500,1,VoicemailMain ;exten => 8500,2,Hangup ; ;
Or a conference room (you'll need to edit meetme.conf to enable this room) ;
;exten => 8600,1,Meetme,1234 ; ; Or playing an announcement to the called
party, as soon it answers ; ;exten =
8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type "show applications"
at
your ; friendly Asterisk CLI prompt.
;
; 'show application <command>' will show details of how you ; use
that
particular application in this file, the dial plan.
;
[from-sip]
exten => 1001,1,Dial(SIP/1001)
exten => 1001,2,Wait(1)
exten => 1001,3,Answer
exten => 1001,4,Hangup
include => demo
include => local
;exten => _9X,1,Dial,Zap/1/${EXTEN:1}
;exten => _9X,2,Goto(102)
;exten => _9X,102,Congestion
;exten => _9X,103,Hangup
ZAPTEL.CONF
loadzone = us
#default = us
fxsks=1
Please, notice that "default=us" is commented since when I run ztcfg
-vvv it
gave me the following error:
root@knoppix:/etc# ztcfg -vvvv
Notice: Configuration file is /etc/zaptel.conf line 2: Unknown keyword
'default'
1 error(s) detected
I commented the line 2 and run ztcfg again and it worked without errors.
May it has no relationship with the error describe above.
Thanks.
Jaime
My zaptel.conf config: -
# Below setting is for E1
span=1,1,0,cas,hdb3
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
My zapata.conf config: -
# Below setting is for E1
switchtype = national
signalling = pri_cpe
group = 1
channel => 1-15
channel => 17-31
My extension.conf config: -
[default]
exten => 181,1,Dial(Zap/1/181)
When I perform a dailing from my SIP Phone, I got the
error message as below: -
-- Executing Dial("SIP/1183000001-6f4e",
"Zap/1/181") in new stack
Aug 22 11:03:02 NOTICE[9023]: app_dial.c:764
dial_exec: Unable to create channel of type 'Zap'
== Everyone is busy/congested at this time
I am beginner...How to solve this?
____________________________________________________
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
Yep, I am connecting to some other equipemnt...its a Clarent gateway equipped with a National Microsystems (Quad Port) --- El Flynn <el_flynn@lanvik-icu.com> wrote:> Hi there, > > Are you getting the E1 span in from Telekom, or are > you connecting to some other > equipment? > > root linux wrote: > > My zaptel.conf config: - > > > > # Below setting is for E1 > > span=1,1,0,cas,hdb3 > > bchan=1-15 > > bchan=17-31 > > dchan=16 > > > > loadzone = us > > defaultzone=us > > > > > > My zapata.conf config: - > > > > # Below setting is for E1 > > switchtype = national > > signalling = pri_cpe > > group = 1 > > channel => 1-15 > > channel => 17-31 > > > > My extension.conf config: - > > > > [default] > > exten => 181,1,Dial(Zap/1/181) > > > > When I perform a dailing from my SIP Phone, I got > the > > error message as below: - > > > > -- Executing Dial("SIP/1183000001-6f4e", > > "Zap/1/181") in new stack > > Aug 22 11:03:02 NOTICE[9023]: app_dial.c:764 > > dial_exec: Unable to create channel of type 'Zap' > > == Everyone is busy/congested at this time > > > > > > I am beginner...How to solve this? > > > > > > > > > > > > > ____________________________________________________ > > Start your day with Yahoo! - make it your home > page > > http://www.yahoo.com/r/hs > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@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> > > > >__________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Nathan C. Smith
2005-Aug-25 07:04 UTC
[Asterisk-Users] Unable to create channel of type 'Zap'
Which side of the span is actually providing the clock? The Asterisk side
or the other side (telco?) If it is the telco the clock needs to be set to
'0'.
Span=1,0,0,cas,hdb3
-Nate
-----Original Message-----
From: root linux [mailto:rootlinux@yahoo.com]
Sent: Sunday, August 21, 2005 10:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Unable to create channel of type 'Zap'
My zaptel.conf config: -
# Below setting is for E1
span=1,1,0,cas,hdb3
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
My zapata.conf config: -
# Below setting is for E1
switchtype = national
signalling = pri_cpe
group = 1
channel => 1-15
channel => 17-31
My extension.conf config: -
[default]
exten => 181,1,Dial(Zap/1/181)
When I perform a dailing from my SIP Phone, I got the
error message as below: -
-- Executing Dial("SIP/1183000001-6f4e",
"Zap/1/181") in new stack
Aug 22 11:03:02 NOTICE[9023]: app_dial.c:764
dial_exec: Unable to create channel of type 'Zap'
== Everyone is busy/congested at this time
I am beginner...How to solve this?
____________________________________________________
Start your day with Yahoo! - make it your home page
http://www.yahoo.com/r/hs
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@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
I installed Asterisk 1.0 CVS on a Debian Sarge System. I am using two
ISDN-HFC-Cards and a point-to-point ISDN Connection.
Everything seemed to work pefectly. But today I realized that I cannot
use two lines at the same time. I get the error message:
3 active channel(s)
asterisk*CLI> show channels
Channel (Context Extension Pri ) State Appl.
Data
Zap/2-1 (internS0 555858 2 ) Ring (None)
(None)
SIP/user1-e4eb (from-sip 1 ) Up Bridged Call
Zap/4-1
Zap/4-1 (externS0 4445858 1 ) Up Dial
SIP/user1|20|t
Sep 27 15:13:56 NOTICE[13491]: app_dial.c:805 dial_exec: Unable to
create channel of type 'Zap'
I restarted Asterisk and it worked for 1 or 2 times. After that I had
the same problem.
Any hints, where I can start searching? Is there any possiblity to
force the channel to "hangup" or something like that?