Okay. Well, I completely agree with you, there should be a way of enabling that for easier transition. However, due to the way the system is designed, I’m afraid it doesn’t appear very easy to do, at least not without modifying some C code.
The optimal solution would be to make sure the SiLabs chipset (that interfaces with the phone line on a TDM400P for ex.) is set up to monitor line voltage (something the chip can do but probably isn’t being used by ZAP driver) and then use a drop in line voltage prior to an “Answer” call to cancel the incoming call instance to Asterisk (thus aborting the call before getting to the Answer instruction). Seperately, there should be a change to the Incoming Routes configuration (in the gui code) to specify an optional pre-answer wait time - which would give something else a chance to pick up before Asterisk did. Would work great for a parallel fax device too.
Another possibility is found at: http://www.voip-info.org/wiki/view/Dialtone+and+line-in-use+detection+on+a+ZAP+channel which describes an extra bit of code that presumably can detect pickup of line - except that it has to pick up the line to do so? This doesn’t sound like it would work right to me.
So, it’s not impossible, but probably not available real soon now either.
Anybody out there with knowledge of ZAP code want to collaborate on adding pre-answer off-hook detection?
-----Original Message-----
From: [email protected]
[mailto:[email protected]]On Behalf Of
lperdue
Sent: Monday, July 10, 2006 03:04
To: [email protected]
Subject: Re: [Amportal-users] Operator Intervention
Yes, you are correct about the analog end of things. I am trying to transition users gradually by duplicating the system we currently have from AT&T voicemail …
I was hoping to hack FreePBX to allow an IF/THEN decision in the incoming message initiation … thus IF rings =4 or >4, THEN answer; IF rings<4 THEN Hang up.
Something like this would speed acceptance of FreePBX and TrixBox because the system viability and features could be demo’ed with the existing phone lines and handsets … this would allow users and management to see that they were functionally equivalent to the existing system … and that would help get approval for IP phones etc. to build the new system out and show how superior it is to the status quo.
scott.griepentrog at t… wrote:
[quote] I did that using a ring group. The incoming call is sent to a specific ring group, which rings the secretary’s phone for say 10 seconds, then defaults out to the regular attendant menu for the company. Incoming callers hear one PSTN ringback, then a few more * ringback, then either the secretary or the menu of options for them to dial.
-----Original Message-----
From: [email protected]
[mailto:[email protected]]On Behalf Of
lperdue
Sent: Thursday, July 06, 2006 05:29
To: [email protected]
Subject: [Amportal-users] Operator Intervention
I need a way for a live operator to have the opportunity answer before FreePBX captures the call … is this done in a from_pstn_custom file by defining the Live Operator as an extension that tells FreePBX to hang up if it does not receive a given number of ring signals?
462 W. Napa St., Suite 201, Sonoma, CA 95476,
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Email: [email protected]
http://www.ideaworx.com
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[/quote]
462 W. Napa St., Suite 201, Sonoma, CA 95476,
Phone: 707-326-4503, fax: 707-940-4146
Email: [email protected]
http://www.ideaworx.com
Using Tomcat but need to do more? Need to support web services, security?
Get stuff done quickly with pre-integrated technology to make your job easier
Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
Amportal-users mailing list
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