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

Jaime Blanco blancolandau at hotmail.com
Wed Apr 20 19:50:29 MST 2005


Thanks Rob, Robert and Tim,

the issue, in fact, the group keyword in zapata.conf.

Best regards.
Jaime

>From: "Tim Touhsaent" <touhsatj at hotmail.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
><asterisk-users at lists.digium.com>
>Subject: Re: [Asterisk-Users] Unable to create channel of type 'Zap'
>Date: Wed, 20 Apr 2005 15:38:31 -0400
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>To help you out i will post my config files... The only problem that i have
>is in an active call i can't get my phones to send responses to voice menus
>such as dialling the voicemailmain cmd.
>
>
>I am using a TDM04b card with four ports instead of one my zapata.conf file
>looks like:
>
>[trunkgroups]
>
>[channels]
>
>;this block is standard features of the anolog lines
>musiconhold=default
>rxwink=300 ; seems to use long winks
>usecallerid=yes
>hidecallerid=no
>callerid="Berkleigh Computer Systems"
>Callwaiting=yes
>busydetect=no
>callprogress=no
>usecallingpres=yes
>callwaitingcallerid=yes
>threewaycalling=yes
>transfer=yes
>cancallforward=yes
>callreturn=yes
>echocancel=yes
>echocancelwhenbridged=yes
>echotraining=800
>rxgain=3.3 ; the gain options are volume adjustments (recieve&transmit)
>txgain=3.3
>
>group=0
>callgroup=1
>pickupgroup=1
>immediate=no
>faxdetect=no
>
>signalling=fxs_ks ;fxo cards use fxs signalling
>callerid=asrecieved
>context=default ; context in extensions.conf to use
>channel=>1-4 ; we have four availlable ports on our card
>
>and my zaptel.conf has:
>
>fxsks=1-4
>defaultzone=us
>loadzone=us
>
>
>the only difference should be that you should have 1 instead of 1-4 in both
>zapata and zaptel. as far as the extensions.conf i had issues with trying 
>to
>work with the demo file and just wrote my own, I found it easier, what you
>prefer is your choice.
>
>Tim Touhsaent
>Berkleigh Computer Systems
>Kutztown, PA
>
>----- Original Message -----
>From: "Jaime Blanco" <blancolandau at hotmail.com>
>To: <asterisk-users at lists.digium.com>
>Sent: Wednesday, April 20, 2005 2:33 PM
>Subject: [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 at 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 at 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 at misery.digium.com/s at 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 at 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 at ximark.com> 
>jaime.blanco at ximark.com
> > Personal email:    <mailto:jaime at blanco.com> jaime at blanco.com
> > MSN ID:               <mailto:blancolandau at hotmail.com>
> > blancolandau at hotmail.com
> >
> >
> > _______________________________________________
> > 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
> >
>_______________________________________________
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