Hello Kerry, Maybe it's me, but why are you using hint in this fashion? Shouldn't you be doing exten => 100,1,Dial(SIP/900&zap/g0/w5551212) or is there something new that I have missed?
Regards, Greg -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Saturday, December 31, 2005 11:38 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Having major issues with TDM2400 To summarize, I spent 6 hours yesterday on the phone with Digium trying to fix a problem with the TDM2400 ad we still don't have it working right. The lastest version of everything are installed and confirmed by Digium. So here is the issue: Zapata.conf ; Disable call progress ; callprogress=yes Outbound calls to PSTN phone numbers work properly But using this: exten => 100,hint,SIP/900&&zap/g0/w5551212 The extension will ring once, but as soon as the PSTN line is picked up, the sip phone stops ringing because * thinks the phone has been answered. Zapata.conf ; Enable call progress callprogress=yes Outbound calls to PSTN phone numbers will dial out but there is no answer detection from the far side. The far side may answer the phone but * keeps ringing until the timeout expires. And using this: exten => 100,hint,SIP/900&&zap/g0/w5551212 Both the sip phone and zap line both ring at the same time until the time. Picking up the sip phone bridges the call and disconnects the zap line as it should. Any ideas? We are stuck until after the holidays at this point. -Kerry _______________________________________________ --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 _______________________________________________ --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