[Asterisk-Users] Having major issues with TDM2400

C F shmaltz at gmail.com
Sun Jan 1 19:23:55 MST 2006


On 1/1/06, Kerry Garrison <support at techdatapros.com> 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: 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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