[Asterisk-Users] Having major issues with TDM2400

Sun Jan 1 18:36:02 MST 2006

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,
their bandwidth is insuffecient for using a SIP provider, although a T1 line
is on order.

-Kerry

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> gw at adcomcorp.com
> Sent: Sunday, January 01, 2006 5:08 PM
> To: asterisk-users at lists.digium.com
> 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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Gregory Wiktor - ADCom Corp.
> Sent: Sunday, January 01, 2006 8:05 PM
> To: asterisk-users at lists.digium.com
> 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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] 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 <support at techdatapros.com> 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
> > >
> > >
> > >
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