[Asterisk-Users] Having major issues with TDM2400

[C F]

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

How? using Zap FXS? or Zap FXO?
The question has been answered by me and BJ, You will not get status
of the POTS using Zap, because it's already answered as soon as you
take it off hook, some good workaround examples exist in the user list
archive, amongst them:
* Implement a macro using the M option in the dial command to not
bridge the call until a certain key is pressed.
* Implement the c option for the zap channel.

Again this is NOT a problem with Digium/TDM2400/Asterisk/Zaptel, but
with you reposting the same question after it has been answered, maybe
you should not use AMP but Asterisk from source then you will
understand this better.

> 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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