On 1/1/06, Kerry Garrison <[EMAIL PROTECTED]> wrote: > Thanks everyone, the reason I posted here was because a Digium support tech > said "it should work" and he couldn't figure it out. So while I appreciate > everyone's comments that it "wont work", a technician from Digium said it > should, hence I turned to the list for clarification. This is not really a > good answer for me to go back to my client with as this is one primary > feature he liked which pushed him into an Asterisk solution. For right now,
It will still work using the M option in the dial command, as I wrote before, also look up the follwoing: http://www.voip-info.org/wiki-asterisk+cmd+dial http://bugs.digium.com/view.php?id=5574 Using some creativity you can give your client what you promised plus. > their bandwidth is insuffecient for using a SIP provider, although a T1 line > is on order. > > -Kerry > > > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > [EMAIL PROTECTED] > > Sent: Sunday, January 01, 2006 5:08 PM > > To: [email protected] > > Subject: RE: [Asterisk-Users] Having major issues with TDM2400 > > > > Oh just a followup, if you are trying to do an outbound > > dialout over analog, what others are saying is correct. You > > could consider however using a voip provider to make the > > outbound call, then you should have status. > > > > Greg > > > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Gregory Wiktor - ADCom Corp. > > Sent: Sunday, January 01, 2006 8:05 PM > > To: [email protected] > > Subject: RE: [Asterisk-Users] Having major issues with TDM2400 > > > > Hello Kerry, I do it exactly as such, however in steps. My > > understanding of the hint system is just for notification of > > status, not for execution of dialing. > > > > I regularly use this same setup you are looking for, rings > > in, then rings 2-5 devices (some zap, some iax) and the first > > one that answers gets the call. > > > > Make sure you use the Dial( command I replied with > > previously. (avoid hint for testing). > > > > Looking at your emails, it looks like you need to review the > > dialplan setup, for example the hint and && do not look right to me. > > > > One example for me: exten => > > s,8,Dial(IAX2/ArdsleySomers/314&IAX2/ArdsleySomers/331,,) > > > > But it is the same as SIP/220&Zap/5, etc. > > > > I cannot say anything specific to amp however. > > > > Greg > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Kerry Garrison > > Sent: Sunday, January 01, 2006 7:34 PM > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: RE: [Asterisk-Users] Having major issues with TDM2400 > > > > 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 > > _______________________________________________ > > --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 > _______________________________________________ --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
