Not using ZAPTEL/DIGIUM for timing

Asterisk Asterisk Support
1

Hi,

Im not comfortable with ztdummy (new to linux). What I want to know is, what are the things which wont work without ZTDUMMY ?? I presume meetme/conference, isnt it?

My purpose is to simply make outbound calls (no transfers etc.) and also to familiarize with the system so that I gradually move to advance configs. So, can I avoid timing concept??

Help shall be appreciated.

btw : im installing VICIDIAL over it…

2

ZTdummy is what you would use to keep the timing for conference / meetme calls if you don’t have a digium card. I don’t have a digium card in my system and use SIP so I need to use ZTdummy. It has worked fine for all my phones / conferences.

3

How do I get it to work?

I have the zaptel tar. Inside is modules/zaptel. I have removed # in front ztdummy in the makefile.
What else to I need to do. Do I need to run make and make install? Do I need to make sure zaptel is loaded somewhere?

I also sort of newbie to Linux (Ubuntu 7.04) so this is tricky :wink:

Rgds

Patrick Arkley

4

well you shouldn’t need to remove the # in the make file. When I build I just do the following:

./configure
make
make install

just make sure you’ve created a sym link to your kernel sources and that you’re using the proper kernel. IE: kernel source = active kernel. Any box that’s hyperthread capable likes to try and get the SMP kernel.

5

I am using SuSE 10.3 (PPC) on a 2.6.22.17-0.1-default kernel. ZTDUMMY was built from source and installed. Still getting “chan_iax2.c: Unable to support trunking on user ‘RioRancho’ without zaptel timing” messages. SIP is working on both boxes, but trunking is not. :cry:

Other box is running AsteriskNOW, asterisk-gui installed on G4MAC to provide web management similar to AsteriskNOW.

One reason I have found in the documentation is that ztdummy requires a UHCI usb controller, where I only have EHCI and OHCI usb controllers. According to that same documentation, any 2.6 kernel makes this un-necessary, which seems to be a mis-statement.

I just want this pbx to support one phone in a remote location. Already have Pap2-T working via SIP. Just wanted to do the <==> thing for the experience. It has been an experience.