I submitted a bug report (http://bugs.digium.com/bug_view_page.php? bug_id=0002620) regarding this issue a couple days ago, and it has since been fixed. You can download the patch from the above link, or wait a bit and it will probably be applied to the stable CVS branch.
On Sat, 2004-10-09 at 23:41 -0400, Jeff owen wrote: > Ok, now since I have inbound working properly the outbound seemed to > have gotten hosed. > > > > In the extensions.conf I have it setup as: > > > > exten => _9NXXXXXXXXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) > > > > In the sip.conf I have it setup as: > > > > [pstn] SPA-3k PSTN Line > > type=friend > > context=default > > secret=supersecretpassword > > port=5061 > > host=dynamic > > dtmfmode=rfc2833 > > canreinvite=no > > nat=no > > > > Which should be correct for inbound and outbound calling, right? > > > > However all I get when I try and dial out is another dial tone and if > I try to dial a number a second time the call will go thru. Kind of > like dialing 98145551212, getting dial tone, then dialing 8145551212 > and the call gets connected then. > > > > I’m not sure what needs to be set on the SPA-3k to allow calls to be > made by what is passed to it. However, when I look at the SIP debug I > don’t even see the number listed as passed to the SPA-3k. > > > > Any ideas on where to look or what to set? > > > > Thanks, > > > > Jeff > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users