Ooops. I misread your question 1. More than likely that is caused by
Asterisk not detecting the *1. Make sure your phone has DTMF set to RFC
mode.
Michael Shuler, C.E.O.
OmniLEC And BitWise Communications, Inc.
[quote] -----Original Message-----
From: [email protected]
[mailto:[email protected]] On
Behalf Of Michael Shuler
Sent: Thursday, August 10, 2006 10:19 PM
To: [email protected]
Subject: Re: [Amportal-users] A few questions
-
If during a barge from HUD you need to enable it under
/etc/asterisk/extensions_hud.conf by changing the [barge] section to:
[barge]
;
; support the ability of barging into calls
;
exten => _1X.,1,MeetMe(20000000${EXTEN:1},qd,271721)
exten => _2X.,1,MeetMe(20000000${EXTEN:1},aqds,271721)
-
Most likely it’s the phone you are calling from but it
could be some
weird settings that you will have to play with for the rest
of your life
until you get lucky. Try a Grandstream GXP-2000 directly to
it, that’s what
we use and everything is crystal clear. We also gateway to
the PSTN via a
PRI which makes calls sound perfect (Assuming PRI or VoIP on
the other end).
-
Sorry, never tried that.
-
Sorry, never tried that yet either.
-
Once it reaches the VoIP carrier its up to them as to if
they are going
to drop it out directly to the PSTN from he gateway you came
in on or if
they are going to ship it around via VoIP, TDM or ATM. Most
of the time it
really wont matter. But every VoIP service provider will be
using digital
lines i.e. PRI to interface to the PSTN, hence pending any Internet
flakeyness your call quality will be better than anything you try with
analog lines at your home or business. As to how much, well
that depends on
how good your analog lines are.
Michael Shuler, C.E.O.
OmniLEC And BitWise Communications, Inc.
-----Original Message-----
From: [email protected]
[mailto:[email protected]] On
Behalf Of Carlos Leal
Sent: Thursday, August 10, 2006 9:51 PM
To: [email protected]
Subject: [Amportal-users] A few questions
OK, Finally got Trixbox 1.1.1 going with FreePBX 2.1.1.
Lots of reading later and knowing there’s more I don’t know
than what
I do know, these are the answers I’m stuck on at present.
If this is
the wrong forum for these questions please advise.
-
Recording Calls - *1 does not seem to do anything. Is there
something that needs to be done to activate it? The extensions are
set to “On Demand.”
-
Recording in general - If the voice mail recordings seem
muddy and
not very loud. What component should I look to. PC hardware with a
clone X101P for PSTN connections and a PAP2 to feed my
home’s phone
lines. The tx/rx values don’t reflect changes made in zapata.conf
when running zttool. Is the card bad?
-
Pattern matching - I took the Blacklist module and used
part of it
to create a White List that filters our evening callers to
only those
on the list. Is there any way to do pattern matching so
that ranges
of numbers can be evaluated in one pass using Asterisk’s built-in
database?
-
Backup/Restore - Does this work on FreePBX 2.1.1 as
intended? I’ve
read conflicting reports.
-
Distinctive Ring - Can it be done by Trunk to an extension
connected to a PAP2? I’d like to know if a call is coming
in is from
a business line or a personal line. I used to subscribe to
a virtual
PBX that would alert me of an incoming business call by playing a
couple of beeps prior to transferring the call. I then knew how to
answer the call.
-
This last one is not FreePBX specific but maybe someone
knows the
answer. At what point does a VoIP call to a landline leave the
internet and get on phone company type circuits? I’m trying to
determine if getting a VoIP provider that’s geographically
close will
increase the odds of getting better call quality, everything else
being equal.
Thanks for reading.
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[/quote]
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
[email protected]
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