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[asterisk-users] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 -


 
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PostPosted: Wed Apr 30, 2008 7:22 am    Post subject: [asterisk-users] Zap channels on TDM400P report to be unimpl Reply with quote

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
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stotaro at totarotechn...
Guest





PostPosted: Wed Apr 30, 2008 7:41 am    Post subject: [asterisk-users] Zap channels on TDM400P report to be unimpl Reply with quote

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:
Quote:


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
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stotaro at totarotechn...
Guest





PostPosted: Wed Apr 30, 2008 8:20 am    Post subject: [asterisk-users] Zap channels on TDM400P report to be unimpl Reply with quote

My question is does ANYONE do ANY testing on these releases? It would
seem that this bug is so paramount to the purpose of the code that had
anyone taken a MINUTE to TEST, it would have been discovered
IMMEDIATELY.

sigh.....

Thanks,
Steve Totaro

On Wed, Apr 30, 2008 at 8:41 AM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:
Quote:
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:
Quote:


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?
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