[asterisk-dev] FXO with Call Forwarding
Patrick Greene
patrick.greene at evot.net
Fri Mar 31 07:31:13 MST 2006
You are correct - our dial plan has an Answer statement followed by a Dial
so that the call rings at the station we want it to. This was how we have
seen several dial plans do this. Can we just use the Dial command
immediately in the dial plan without an Answer first so that it continues to
ring both on the PSTN line as well as the phone? I didn't think you could.
I'm new to this forgive me if this is a dumb question.
Patrick Greene
610-798-4896
-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Rich Adamson
Sent: Friday, March 31, 2006 9:11 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] FXO with Call Forwarding
> We are using Asterisk 1.2 and have a TDM 400P installed with 2 FXO ports
> - each of these is plugged into a separate PSTN line with its own phone
> number. We have everything configured pretty well, but one strange issue
> is occurring which I'm hoping you can shed some light on. We use call
> forwarding from the PSTN provider (Verizon) when we are going to be
> away. In order to turn on call forwarding, we grab an outside line and
> dial *72 and then the # we want to forward to. To disable call
> forwarding we dial *73. After we have forwarded our calls through
> Verizon, when a call rings at the forwarded line, we get a single ring
> in Asterisk before the call is forwarded. This behavior is expected as
> it happened with the old phone system. However, Asterisk doesn't seem to
> drop that channel after that call rings in and it will remain in the
> "Off Hook" state until pulling out the phone line and plugging it back
> in to Asterisk. Is there a reason for this?
Asterisk (and the associated TDM400 drivers) will not answer that
ringing line unless you have something in your config that tells it to
answer. That could range from having an "answer" statement in your
dialplan to an improperly configured ivr, and possibly incorrect
zapata.conf parameters.
Without knowing what those sections of zapata.conf and extensions.conf
look like, its anybodies guess what might be wrong.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
More information about the asterisk-dev
mailing list