[Asterisk-Users] problems with X100P - No channeltyperegisteredfor 'Zap'

Wiley E. Siler wsiler at e2020inc.com
Tue Oct 5 00:48:01 MST 2004


Just to make sure this isn't a typo in your original email...  Is this
example from your zapata.conf?

Also, the extension you have shown are in extensions.conf not
zapata.conf correct?

Here is an example of a good zapata.conf....

[channels]
language=en
busydetect=yes
faxdetect=both
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
useincomingcalleridonzaptransfer=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=1.5
txgain=1.5
group=1 
signalling=fxs_ks
callerid=asreceived
context=incoming
channel=>1-8

These ones above are the only values I need in mine.  The context and
channels are the last entries.

Next.... Zaptel.conf located in /etc/ has these entries only...

fxsks=1-8
loadzone=us

This sets my 1-8 channels to the correct type of channel.  Fxs
kewlstart, note that zapata has a line called signalling that matches.

Now finally extensions.conf

[incoming] 
include => mainmenu 
include => sip

Sip is my context with my internal phones
Mainmenu is my context with my autoattendant settings

Check the wiki on extensions.conf, zapata.conf, and zaptel.conf

Regards,
Wiley


 

-----Original Message-----
From: khoonking [mailto:khoonkingsg at yahoo.com.sg] 
Sent: Tuesday, October 05, 2004 12:15 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] problems with X100P - No
channeltyperegisteredfor 'Zap'

Hi,

I have confirmed that there is no space between "=>" but still the same
problem.

> zapatel.conf
> ========================
> [channels]
> signalling=fxs_ks
> context=incoming
 channel => 1 ;X100P

>[incoming]
>;exten => s,1,Echo ;for testing the connection
>exten => s,1,Answer			; Answer the line
>exten => s,2,Playback,demo-thanks ;for playing a file

> [default]
> exten => 6200,1,Dial(SIP/6200,20)
> exten => 6203,1,Dial(SIP/6203,20)
> exten => _X.,1,Dial,Zap/1/${EXTEN}

When I call from an analog phone to the X100P ext, Asterisk CLI screen
does not print anything. 
Based on my configuration, it should answer the call automatically and
playback demo-thanks.

When I tried to dial out using the SIP IP-Phones, I am getting

Executing Dial("SIP/6201-f848", "Zap/1") in new stack Oct  5 15:01:41
WARNING[15375]: channel.c:1901 ast_request: No channel type registered
for 'Zap'
Oct  5 15:01:41 NOTICE[15375]: app_dial.c:742 dial_exec: Unable to
create channel of type 'Zap'
  == Everyone is busy/congested at this time


Anyway way of ckecking that my Zap channel is working? 

 
Meng Kim
 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven
Critchfield
Sent: Tuesday, October 05, 2004 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] problems with X100P - No
channeltyperegistered for 'Zap'

Don't group reply. I get mail just fine from the mailing list like
everyone else. The mailing list is properly set up to promote community
by helping keep discussions on the list for all to see and benefit.

On Tue, 2004-10-05 at 14:23 +0800, khoonking wrote:
> Hi,
> 
> This is what I have configured. I have connected 2 sip IP-Phones to 
> the asterisk. SIP IP-Phone 1 can call SIP IP-Phone 2. The X100P is 
> connected
to
> my PBX ext 100. 
> I would like my SIP IP-Phones to be able to call to my office colleage
ext,
> e.g. 101
> And my IP-Phones to receive incoming call from PBX.
> 
> 
> Zaptel.conf
> ====================	
> fxsks=1 #X100P
> loadzone = us
> defaultzone=us
> 
> 
> zapatel.conf
> ========================
> [channels]
> signalling=fxs_ks
> context=incoming
> channel = >1 ;X100P

Verify there isn't a space in between the equals sign and the greater
than sign. I am betting you are getting a parse error there and it is
causing you the grief. Also add a group there so your system is prepared
for growth. It is good form too.

> extension.conf
> 
> [default]
> exten => 6200,1,Dial(SIP/6200,20)
> exten => 6203,1,Dial(SIP/6203,20)
> exten => _X.,1,Dial,Zap/1/${EXTEN}

once you have a group defined above, this changes to g1 or whatever
group number you define. Again good form in case you add a second line,
you only have to configure the line into the group and your extensions
file is good to go.


--
Steven Critchfield <critch at basesys.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

_______________________________________________
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


The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer




More information about the asterisk-users mailing list