[Asterisk-Users] Having major issues with TDM2400

[Kerry Garrison]

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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