On Thu, Oct 30, 2008 at 12:56:06PM -0700, hin lee wrote: > Tzafrir, > > You are correct! I didn't have to commented out the unused FXO ports. So to > revise my earlier email, I have to do the following: > > 1) Run genzaptelconf > > 2) Run "cat /proc/zaptel/*" to find the channel my line is connected to. > > 3) Add my channel to /etc/asterisk/zapata-channels.conf
Why is that? genzaptelconf should generate that file. > > ie. channel => 1 > > I'm not sure why I have to do this manually. My zapata-channels.conf file > is blank and doesn't work until I put the "channel => X" to it. > > 4) Of course, reboot the server. Why is that? 'dahdi restart' should do. Or in the worst case, restart asterisk. > > > > > --- On Thu, 10/30/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > > From: Tzafrir Cohen <[EMAIL PROTECTED]> > > Subject: Re: [asterisk-users] CHANUNAVAIL with a TDM800 card > > To: [email protected] > > Date: Thursday, October 30, 2008, 11:20 AM > > On Thu, Oct 30, 2008 at 11:03:03AM -0700, hin lee wrote: > > > I got this working. For what it's worth, > > here's what the issue. > > > > > > The channel wasn't getting created under FreePBX > > via script. Here's what I needed to do: > > > > > > 1) Run genzaptelconf to generate the zaptel configs > > > > This generates you /etc/zaptel.conf and > > /etc/asterisk/zapata-channels.conf (or > > /etc/asterisk/zapata-auto.conf , > > in the modified versions by some distributions). > > > > > > > > 2) find the channel the port(s) is on. > > > > > > cat /proc/zaptel/* > > > > > > 3) comment out the unused ports in /etc/zaptel.conf > > based on step 2 result. > > > > Why? > > > > > > > > 4) put in the available channel in > > /etc/asterisk/zapata-channels.conf > > > > > > ie. channel => 1 > > > > So I gather it has generated for you zapata-auto.conf but > > zapata.conf > > #include-s zapata-channels.conf . You're confused (or > > someone did some > > bad integration work). > > > > > > > > 5) comment out the unused channels in > > /etc/asterisk/zapata-auto.conf > > > > > > > > > > > http://www.freepbx.org/support/documentation/administration-guide/interfacing-to-a-pstn > > > > This thing should be removed. It is aufully confusing and > > completely > > outdated. I think it is slightly worse than no information > > at all, as > > the defaults of freepbx would have worked for you if you > > just ran > > genzaptelconf . > > > > -- > > Tzafrir Cohen > > icq#16849755 jabber:[EMAIL PROTECTED] > > +972-50-7952406 mailto:[EMAIL PROTECTED] > > http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir > > > > _______________________________________________ > > -- 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 > > > > > _______________________________________________ > -- 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 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir _______________________________________________ -- 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
