The goal is to create a user that has a SIP device and a custom ZAP channel device, have them both ring until one is answered, basically a ring group. But I am using AMP's users and device mode rather than the extensions mode. I have this working properly on my office system. However, with the TDM2400 I cannot have both the zap channel and sip channel ringing at the same time and only handing the call to the end device that answers the call. I don't understand why this is so difficult for everyone to grasp. Send a call to both a custom ZAP device and a sip phone and whoever answers it gets the call. -Kerry
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of C F > Sent: Sunday, January 01, 2006 4:14 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Having major issues with TDM2400 > > On 12/31/05, Kerry Garrison <[EMAIL PROTECTED]> wrote: > > 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 > > What are you trying to do here? You trying to hint to a zip > channel and dial a number using the hint priority? > > > > > 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. > > Which makes sense to me, since as soon as you start dialing > you *are* off hook, which in analog means the phone *is* > answered. Since all the singalling is done in band, it is not > difference than picking up the Zap channel for incoming call, > at which point you also understand it's considered 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. > > > > So don't use callprogress if it doesn't work for you, in no > way do I see this related to the subject line of this post. > > > And using this: > > > > exten => 100,hint,SIP/900&&zap/g0/w5551212 > > > > Again what is this suppose to do? > > > 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 > _______________________________________________ --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
