thr3ads.net - asterisk users - [asterisk-users] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error) [Apr 2008]

If this information is useful, please help other people find it:
Share via:

Zoran Milenkovic, Datatek d.o.o.

2008-Apr-30 12:22 UTC

[asterisk-users] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)

Hi list!



I have RHEL Server rel5 (Tikanga)  2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21 EST
2007 i686 i686 i386 GNU/Linux
with installed digium packets

1. Asterisk 1.4.19
2. Zaptel 1.4.10
3. Libpri 1.4.3



My Digium hardware is

[root at asterisk ~]# zaptel_hardware
pci:0000:04:00.0     wctdm+       e159:0001 Wildcard TDM400P REV I

...and I have 2 FXO and 2 FXS ports installed inside the TDM400P card



The problem is the asterisk doesn't recognize the Zap channels at all. The
error is "No channel type registered for 'Zap'
" and "Unable to create channel of type 'Zap' (cause 66 -
Channel not implemented)" and there is the original output form Astersik
console:

    -- Executing [12 at local:1] Dial("SIP/zoran-09f1bf90",
"Zap/3|20") in new stack
[Apr 30 13:17:48] WARNING[2845]: channel.c:3001 ast_request: No channel type
registered for 'Zap'
[Apr 30 13:17:48] WARNING[2845]: app_dial.c:1183 dial_exec_full: Unable to
create channel of type 'Zap' (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [12 at local:2] Hangup("SIP/zoran-09f1bf90",
"") in new stack
  == Spawn extension (local, 12, 2) exited non-zero on
'SIP/zoran-09f1bf90'


And everything was working quite fine when I was on asterisk 1.2.13, previously
installed on this very same server, same Digium card etc.

The configurations are totaly the same, also.

What could be the resolution of this problem?

Here are my configs

[root at asterisk ~]# cat /etc/zaptel.conf
fxsks=1
fxsks=2
fxols=3
fxols=4

[root at asterisk ~]# cat /etc/asterisk/zapata.conf
[channels]
context=incoming
callerid=yes
hidecallerid=no
imidiate=no

context=incoming
signalling=fxs_ks
echocancel=yes
group=1
channel => 1
channel => 2

context=local
signalling=fxo_ks
echocancel=yes
group=2
channel => 3
channel => 4

[root at asterisk ~]# cat /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[mobile]
exten => _906NXXXXXXX,1,Dial(Zap/1/${EXTEN:1})
exten => _906NXXXXXXX,2,Hungup()

[outbound]
exten => _9ZXXXXX.,1,Dial(Zap/1/${EXTEN:1})
exten => _9ZXXXXX.,2,Hangup()

[voicemail]
exten => 31,1,VoiceMailMain(1010 at mail_box)

exten => 33,1,VoiceMailMain(3030 at mail_box)

[konferencija]
exten => 40,1,Meetme(40,s)
exten => 40,2,Hangup()

[interno]
exten => 21,1,Dial(SIP/maja,20)
exten => 21,2,Hangup()

exten => 24,1,Dial(SIP/esad,20)
exten => 24,2,Hangup()

[local]
exten => 11,hint,SIP/cisco1
exten => 11,1,Dial(SIP/cisco1,20)
exten => 11,2,Hangup()

exten => 12,hint,Zap/3
exten => 12,1,Dial(Zap/3,20)
exten => 12,2,Hangup()

exten => 13,hint,SIP/sipura
exten => 13,1,Dial(SIP/sipura,20)
exten => 13,2,Hangup()

exten => 14,hint,SIP/goran
exten => 14,1,Dial(SIP/goran,20)
exten => 14,2,Hangup()

exten => 15,hint,SIP/bobana
exten => 15,1,Dial(SIP/bobana,20)
exten => 15,2,Hangup()

exten => 16,hint,SIP/miroslav
exten => 16,1,Dial(SIP/miroslav,20)
exten => 16,2,Hangup()

exten => 17,hint,SIP/pop
exten => 17,1,Dial(SIP/pop,20)
exten => 17,2,Hangup()

exten => 18,hint,SIP/zoran
exten => 18,1,Dial(SIP/zoran,20)
exten => 18,2,Hangup()

exten => 20,hint,SIP/dusan
exten => 20,1,Dial(SIP/dusan,20)
exten => 20,2,Hangup()

include => outbound
include => mobile
include => konferencija
include => voicemail

[incoming]
exten => 11,1,Dial(SIP/cisco1,20)
exten => 11,2,VoiceMail(1010 at mail_box)
exten => 11,3,Playback(vm-goodbye)
exten => 11,4,Hangup()
exten => 11,102,VoiceMail(1010 at mail_box)
exten => 11,103,Hangup()

exten => 12,1,Dial(Zap/3,20)
exten => 12,2,Playback(vm-goodbye)
exten => 12,3,Hangup()
exten => 12,102,Playback(tt-allbusy)
exten => 12,103,Hangup()

exten => 13,1,Dial(SIP/sipura,20)
exten => 13,2,VoiceMail(3030 at mail_box)
exten => 13,3,Playback(vm-goodbye)
exten => 13,102,VoiceMail(3030 at mail_box)
exten => 13,103,Hangup()

exten => 14,1,Dial(SIP/zoran,20)
exten => 14,2,VoiceMail(3030 at mail_box)
exten => 14,3,Playback(vm-goodbye)
exten => 14,102,VoiceMail(3030 at mail_box)
exten => 14,103,Hangup()

exten => 15,1,Dial(SIP/rzoran,20)
exten => 15,2,VoiceMail(3030 at mail_box)
exten => 15,3,Playback(vm-goodbye)
exten => 15,102,VoiceMail(3030 at mail_box)
exten => 15,103,Hangup()

exten => 17,1,Dial(SIP/pop,20)
exten => 17,2,VoiceMail(3030 at mail_box)
exten => 17,3,Playback(vm-goodbye)
exten => 17,102,VoiceMail(3030 at mail_box)
exten => 17,103,Hangup()

exten => 20,1,Dial(SIP/dusan,20)
exten => 20,2,VoiceMail(dusan at datatek.co.rs)
exten => 20,3,Playback(vm-goodbye)
exten => 20,102,VoiceMail(dusan at datatek.co.rs)
exten => 20,103,Hangup()

exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(incoming,s,1)

exten => s,1,Answer()
exten => s,2,Playback(enter-ext-of-person)
exten => s,3,Hangup()

exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()

include => voicemail
include => konferencija

--
Best regards
Zoran Milenkovic
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080430/44b2158d/attachment.htm

Steve Totaro

2008-Apr-30 12:41 UTC

head link

[asterisk-users] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)

Sean Bright to Asterisk
	
show details 4:47 PM (15 hours ago)
	
There is a bug in 'make install' in Zaptel 1.4.10 that causes the
devices to not be installed correctly.  You can either install 1.4.9 or
wait for 1.4.11 to be released.

On Wed, Apr 30, 2008 at 8:22 AM, Zoran Milenkovic, Datatek d.o.o.
<zoran at datatek.co.yu> wrote:>
>
> Hi list!
>
>
>
> I have RHEL Server rel5 (Tikanga) 2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21
> EST 2007 i686 i686 i386 GNU/Linux
> with installed digium packets
>
> 1. Asterisk 1.4.19
> 2. Zaptel 1.4.10
> 3. Libpri 1.4.3
>
>
>
> My Digium hardware is
>
> [root at asterisk ~]# zaptel_hardware
> pci:0000:04:00.0     wctdm+       e159:0001 Wildcard TDM400P REV I
>
> ...and I have 2 FXO and 2 FXS ports installed inside the TDM400P card
>
>
>
> The problem is the asterisk doesn't recognize the Zap channels at all.
The
> error is "No channel type registered for 'Zap'
> " and "Unable to create channel of type 'Zap' (cause 66 -
Channel not
> implemented)" and there is the original output form Astersik console:
>
>     -- Executing [12 at local:1] Dial("SIP/zoran-09f1bf90",
"Zap/3|20") in new
> stack
> [Apr 30 13:17:48] WARNING[2845]: channel.c:3001 ast_request: No channel
type
> registered for 'Zap'
> [Apr 30 13:17:48] WARNING[2845]: app_dial.c:1183 dial_exec_full: Unable to
> create channel of type 'Zap' (cause 66 - Channel not implemented)
>   == Everyone is busy/congested at this time (1:0/0/1)
>     -- Executing [12 at local:2] Hangup("SIP/zoran-09f1bf90",
"") in new stack
>   == Spawn extension (local, 12, 2) exited non-zero on
'SIP/zoran-09f1bf90'
>
>
> And everything was working quite fine when I was on asterisk 1.2.13,
> previously installed on this very same server, same Digium card etc.
>
> The configurations are totaly the same, also.
>
> What could be the resolution of this problem?
>
> Here are my configs
>
> [root at asterisk ~]# cat /etc/zaptel.conf
> fxsks=1
> fxsks=2
> fxols=3
> fxols=4
>
> [root at asterisk ~]# cat /etc/asterisk/zapata.conf
> [channels]
> context=incoming
> callerid=yes
> hidecallerid=no
> imidiate=no
>
> context=incoming
> signalling=fxs_ks
> echocancel=yes
> group=1
> channel => 1
> channel => 2
>
> context=local
> signalling=fxo_ks
> echocancel=yes
> group=2
> channel => 3
> channel => 4
>
> [root at asterisk ~]# cat /etc/asterisk/extensions.conf
> [general]
> static=yes
> writeprotect=no
> autofallthrough=yes
> clearglobalvars=no
> priorityjumping=no
>
> [mobile]
> exten => _906NXXXXXXX,1,Dial(Zap/1/${EXTEN:1})
> exten => _906NXXXXXXX,2,Hungup()
>
> [outbound]
> exten => _9ZXXXXX.,1,Dial(Zap/1/${EXTEN:1})
> exten => _9ZXXXXX.,2,Hangup()
>
> [voicemail]
> exten => 31,1,VoiceMailMain(1010 at mail_box)
>
> exten => 33,1,VoiceMailMain(3030 at mail_box)
>
> [konferencija]
> exten => 40,1,Meetme(40,s)
> exten => 40,2,Hangup()
>
> [interno]
> exten => 21,1,Dial(SIP/maja,20)
> exten => 21,2,Hangup()
>
> exten => 24,1,Dial(SIP/esad,20)
> exten => 24,2,Hangup()
>
> [local]
> exten => 11,hint,SIP/cisco1
> exten => 11,1,Dial(SIP/cisco1,20)
> exten => 11,2,Hangup()
>
> exten => 12,hint,Zap/3
> exten => 12,1,Dial(Zap/3,20)
> exten => 12,2,Hangup()
>
> exten => 13,hint,SIP/sipura
> exten => 13,1,Dial(SIP/sipura,20)
> exten => 13,2,Hangup()
>
> exten => 14,hint,SIP/goran
> exten => 14,1,Dial(SIP/goran,20)
> exten => 14,2,Hangup()
>
> exten => 15,hint,SIP/bobana
> exten => 15,1,Dial(SIP/bobana,20)
> exten => 15,2,Hangup()
>
> exten => 16,hint,SIP/miroslav
> exten => 16,1,Dial(SIP/miroslav,20)
> exten => 16,2,Hangup()
>
> exten => 17,hint,SIP/pop
> exten => 17,1,Dial(SIP/pop,20)
> exten => 17,2,Hangup()
>
> exten => 18,hint,SIP/zoran
> exten => 18,1,Dial(SIP/zoran,20)
> exten => 18,2,Hangup()
>
> exten => 20,hint,SIP/dusan
> exten => 20,1,Dial(SIP/dusan,20)
> exten => 20,2,Hangup()
>
> include => outbound
> include => mobile
> include => konferencija
> include => voicemail
>
> [incoming]
> exten => 11,1,Dial(SIP/cisco1,20)
> exten => 11,2,VoiceMail(1010 at mail_box)
> exten => 11,3,Playback(vm-goodbye)
> exten => 11,4,Hangup()
> exten => 11,102,VoiceMail(1010 at mail_box)
> exten => 11,103,Hangup()
>
> exten => 12,1,Dial(Zap/3,20)
> exten => 12,2,Playback(vm-goodbye)
> exten => 12,3,Hangup()
> exten => 12,102,Playback(tt-allbusy)
> exten => 12,103,Hangup()
>
> exten => 13,1,Dial(SIP/sipura,20)
> exten => 13,2,VoiceMail(3030 at mail_box)
> exten => 13,3,Playback(vm-goodbye)
> exten => 13,102,VoiceMail(3030 at mail_box)
> exten => 13,103,Hangup()
>
> exten => 14,1,Dial(SIP/zoran,20)
> exten => 14,2,VoiceMail(3030 at mail_box)
> exten => 14,3,Playback(vm-goodbye)
> exten => 14,102,VoiceMail(3030 at mail_box)
> exten => 14,103,Hangup()
>
> exten => 15,1,Dial(SIP/rzoran,20)
> exten => 15,2,VoiceMail(3030 at mail_box)
> exten => 15,3,Playback(vm-goodbye)
> exten => 15,102,VoiceMail(3030 at mail_box)
> exten => 15,103,Hangup()
>
> exten => 17,1,Dial(SIP/pop,20)
> exten => 17,2,VoiceMail(3030 at mail_box)
> exten => 17,3,Playback(vm-goodbye)
> exten => 17,102,VoiceMail(3030 at mail_box)
> exten => 17,103,Hangup()
>
> exten => 20,1,Dial(SIP/dusan,20)
> exten => 20,2,VoiceMail(dusan at datatek.co.rs)
> exten => 20,3,Playback(vm-goodbye)
> exten => 20,102,VoiceMail(dusan at datatek.co.rs)
> exten => 20,103,Hangup()
>
> exten => i,1,Playback(pbx-invalid)
> exten => i,2,Goto(incoming,s,1)
>
> exten => s,1,Answer()
> exten => s,2,Playback(enter-ext-of-person)
> exten => s,3,Hangup()
>
> exten => t,1,Playback(vm-goodbye)
> exten => t,2,Hangup()
>
> include => voicemail
> include => konferencija
>
>
> --
> Best regards
> Zoran Milenkovic
> _______________________________________________
>  -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>  asterisk-users mailing list
>  To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

asterisk users - Apr 2008 - Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)

[asterisk-users] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)

[asterisk-users] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)