[Asterisk-Users] Having major issues with TDM2400

C F shmaltz at gmail.com
Mon Jan 2 11:49:16 MST 2006 

I believe that the Mediatrix 1204 also bridges the call as soon as it
is done dialing (you hear ringing from the POTS not from the SIP
channel).

On 1/1/06, Kerry Garrison <support at techdatapros.com> wrote:
> Well, it would have to be 4 of them for each of the available PSTN lines.  I
> have also considered a Mediatrix channel bank.
> -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 7:53 PM
> > To: asterisk-users at lists.digium.com
> > Subject: RE: [Asterisk-Users] Having major issues with TDM2400
> >
> > Perhaps a Sipura-3000 could be of use here? Anyone have any
> > ideas about that?
> >
> > 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 10:39 PM
> > To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: RE: [Asterisk-Users] Having major issues with TDM2400
> >
> > As much as I like the option of implementing a follow-me type
> > of script, the second problem is that the client wants to use
> > AMP to manage the extensions.
> > Just doesn't seem like I have a solution that fits all of the
> > client's requirements. The easiest solution seems to be to
> > use a SIP trunk for the outbound 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 6:24 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] Having major issues with TDM2400
> > >
> > > 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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