[Asterisk-Users] Having major issues with TDM2400

[Kerry Garrison]

The magic setting is callprogress=yes, however, we have this working
properly in the lab but not at this particular client location right now.
Strange, but true.
-Kerry
 

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> steve at daviesfam.org
> Sent: Sunday, January 01, 2006 10:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Having major issues with TDM2400
> 
> 
> 
> On Sun, 1 Jan 2006, Kerry Garrison 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 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
> 
> By default, analogue zap trunks do not have call progress detection.
> 
> Therefore, as soon as the number has been dialled out on the 
> trunk, Asterisk regards the line as being answered.
> 
> Then, the dial command says: "aha, zap/g0/... is answered. 
> Stop ringing the other lines".
> 
> If you are in a supported country (ie basically north 
> america), callprogress=yes in zapata.conf may help.
> 
> But essentially what you are trying to do is not supported on 
> analogue trunks.
> 
> Steve
> 
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