On 1/9/06, Dan Littlejohn <[EMAIL PROTECTED]> wrote: > I have fixed this before, but I cannot for the life of me remember how I did > it. > > I have a TDM400P with one fxo module and one fxs module. I setup > Asterisk @Home and everything works fine, except when I try and call > out. If I call out with a SIP phone it calls the zap extension and > not the pstn line? If I take the zap extension offhook and call with > the SIP phone it dials out the pstn line fine. I am not sure why the > zap extension is being included in the group, but I cannot find where > to change it in AMP or the .conf files. Any help would be > appreciated. > > ; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" > signalling=fxo_ks > ; Note: this is an extension. Create a ZAP extension in AMP for Channel 1 > context=from-internal > group=1 > channel => 1 > > ; channel 2, WCTDM, inactive. > ; channel 3, WCTDM, inactive. > signalling=fxs_ks > ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 > context=from-pstn > group=0 > channel => 4 > > > Dan >
Bad form to post to your own message, but I figured it out and thought anyone else interested would want to know. It is apparently some problem with how the zapata module loads. I switched the fxs and fso lines and all is well. Maybe something is inherited, etc. Seems like a bug to me. signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 context=from-pstn group=0 channel => 4 ; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" signalling=fxo_ks ; Note: this is an extension. Create a ZAP extension in AMP for Channel 1 context=from-internal group=1 channel => 1 Dan _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
