<div>Thanks for your reply. Yes, I do have zaptel conf configured however since there is no zaptel hardware in this server I am confused as to what to setup.</div> <div> </div> <div>I bought a new Grandstream 2000GPX phone and it's really slick. Don't let anyone kid you about Grandstream being a POS phone. This new 200GPX is just what someone on a budget needs and it works. I brought it here to the office and setup three of the lines to use the local * server which is the new one I just built. The other line I setup to use my * server at home. I was pleased to see that when I touched line 2 my * server at home responded and let me have full functionality with it.</div> <div> </div> <div>The problem is the with the conference rooms on the office server. When I dial them either locally or come in from the POTS using my 800 number it answers the calls and works fine but the conference rooms don't work like my home system where the operator asks for the users name
and then announces each entrant in the conference room.</div> <div> </div> <div>Here is my /usr/local/etc/zaptel.conf file:</div> <div> </div> <div>==============</div> <div>fxsks=1<BR>loadzone=us<BR>defaultzone=us<BR>==============</div> <div> </div> <div>Again, I don't know what I'm doing using fxsks=1 because there is no real zap channel in this machine. But I seem to recall having this working with just ztdummy before I added the X101P card to my * home server.</div> <div> </div> <div>Here is my /usr/local/etc/asterisk/zapata.conf</div> <div> </div> <div>==================================</div> <div>[channels]<BR>language=en<BR>context=default<BR>signalling=fxs_ks<BR>channel=>1<BR>disconnectsupervision=yes<BR>usecallerid=yes<BR>hidecallerid=no<BR>callwaiting=yes<BR>threewaycalling=no<BR>transfer=yes<BR>echocancel=yes<BR>echotraining=yes<BR>echocancelwhenbridged=yes</div> <div>==================================</div>
<div> </div> <div>Without any real hardware am I just barking up the wrong tree or do I just need to tweak these files a little.</div> <div> </div> <div>One note though, the performance over this DSL line at the office is a little choppy. I don't know if I'm going to be able to make this work well enough to get the bosses to sign off on it. Comcast does not service this area. I was looking to have cable modem installed here. My connection at home is cable and it runs more than twice as fast as this DSL here.</div> <div> </div> <div><BR><B><I>Richard <r.neese@gmail.com></I></B> wrote:</div> <BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">did you setup your zaptel.conf. what does zap show channels show ?<BR><BR><BR>On Monday 08 October 2007 20:32:19 Frank Griffith wrote:<BR>> When I first started out with * I was not using the X101P card that I have<BR>> installed now. I just setup
another test box at the office and we were<BR>> testing out the conference room capabilities of the server. But we found<BR>> that it needed ztdummy loaded, okay so I fixed that.<BR>><BR>> Still when we call the conference room extension we are not given the<BR>> usual introduction message which asks for the caller's name. Instead we got<BR>> this message on the cli monitor:<BR>><BR>> Oct 8 16:16:15] WARNING[695]: channel.c:3232 ast_request: No channel<BR>> type registered for 'zap' [Oct 8 16:16:15] WARNING[695]: app_meetme.c:2435<BR>> find_conf: No Zap channel available for conference, user introduction<BR>> disabled (is chan_zap loaded?) [Oct 8 16:17:00] WARNING[695]:<BR>> app_meetme.c:2435 find_conf: No Zap channel available for conference, user<BR>> introduction disabled (is chan_zap loaded?)<BR>><BR>> We were able to conduct the conference room calls but no one was asked to<BR>> enter names and no names were announced
when the caller entered the<BR>> conference room. Did I miss something? I thought that I had made this work<BR>> before without needing zaptel hardware and just using ztdummy.<BR>><BR>> Any advice would help.<BR>><BR>><BR>> ---------------------------------<BR>> Shape Yahoo! in your own image. Join our Network Research Panel today!<BR><BR><BR><BR>_______________________________________________<BR>--Bandwidth and Colocation Provided by http://www.api-digital.com--<BR><BR>Asterisk-BSD mailing list<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-bsd<BR></BLOCKQUOTE><BR><p> 
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