[Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)

Tue Nov 25 12:24:46 MST 2003

Billy,

Thanks for the observations.  The caller ID on outbound is unfortunately
part of the way the Manager interface does outbound calling.  Since it
establishes a call from Asterisk to YOU, then establishes another call
from Asterisk to CALLED PARTY then bridges YOU and CALLED PARTY, you are
not actually the "caller" in the call to VM.

The Hold issue is also one I have yet to find my way around.  The
manager interface doesn't have a "Hold" method (yet).  I may very well
hack one into the manager code so that you can Hold and Reconnect calls,
(plus know when your call has been held/reconnected).

Another solution to this for SIP is to use the REFER methods to direct
the UA (in your case the GS Budgetone).  I don't know which UAs support
this.  Considering the limits of the GS phone, I doubt it would.

I am working on a new version of Call Manager and I will have it ready
sometime after the Thanksgiving holiday.

Thanks,

Steve

> -----Original Message-----
> From: Billy Huddleston [mailto:billy at nxs.net]
> Sent: Monday, November 24, 2003 12:45 PM
> To: ssokol at sokol-associates.com
> Subject: Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1
> (Alpha)
> 
> Been trying out your call manager.. Nice work!  Have a few
opservations to
> tell you...
> 
> When dialing out, it doesn't do callerid, I found this out, because I
use
> callerid to know what extension has called voicemail, so all they have
to
> enter in is thier pass code.
> 
> Hold doesn't work with GS Budgetone at all, and, I'm not exactly sure
what
> the deal is with transfer, I can't get it to work..
> 
> Thanks, Billy
> 
> 
> 
> ----- Original Message -----
> From: "Steven Sokol" <ssokol at sokol-associates.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Saturday, November 22, 2003 11:21 AM
> Subject: RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1
> (Alpha)
> 
> 
> > Zoa,
> >
> > When the boxes are red, that usually indicates that the channel is
busy.
> > In the screen shots I sent earlier you will see that one of the
buttons
> > is red:
> >
> > http://www.sokol-associates.com/images/AstMgr.jpg
> >
> > Notice that only the station marked "Test Xten" is red.  This
station is
> > busy (on another call).  I don't know if that has anything to do
with
> > your issue, but I thought I would throw that out.
> >
> > The message you reference below is a "Status" message.  In this
program
> > the Status messages really only serve as keep-alives.  Every 30
seconds
> > the system issues a command "Action:Status" to keep NATs from
closing
> > the connection due to lack of traffic.
> >
> > Try this:  open the command window and try manually executing some
of
> > the CLI commands.  Try "sip show peers" to make sure the SIP peers
are
> > registered.  Also try "sip show channels" to see if there is already
a
> > call terminated at the channel you are calling.
> >
> > I will try to diagnose this further if you can send some additional
> > information.  Please include the monitor.conf file, and if possible
a
> > -vvvv trace from Asterisk.
> >
> > Thanks,
> >
> > Steve
> >
> > > -----Original Message-----
> > > From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-
> > > admin at lists.digium.com] On Behalf Of zoa
> > > Sent: Saturday, November 22, 2003 4:13 AM
> > > To: asterisk-users at lists.digium.com
> > > Subject: RE: [Asterisk-Users] Asterisk Call Manager for Windows
0.0.1
> > > (Alpha)
> > >
> > > [201]
> > > Username=davy
> > > Technology=SIP
> > > DeviceID=davy
> > >
> > > [202]
> > > Username=pieter
> > > Technology=SIP
> > > DeviceID=pieter
> > >
> > > 201 is the extension from in extensions.conf
> > > davy = the thing between brackets in sip.conf
> > >
> > > When i try to click on one of the red boxes in the manager, i
always
> > get:
> > >
> > > Event: Status -|- Channel: SIP/davy-1a07 -|- CallerID: davy -|-
State:
> > Up
> > > -|- Context: sip -|- Extension: 202 -|- Priority: 1 -|- Link:
> > > SIP/pieter-e582 -|- Uniqueid: 1069581.102
> > > Response: Error -|- Message: Invalid channel
> > > At 21:06 21/11/2003 -0600, you wrote:
> > > >Here's the structure for the monitor.conf file:
> > > >
> > > >[1101]                  Extension Number (from extensions.conf in
> > > >Asterisk)
> > > >UserName=Blah Blah      Label.  Simply sets the caption for the
> > button.
> > > >Technology=SIP          Technology used for stations (SIP, MGCP,
Zap,
> > > >etc.)
> > > >DeviceID=1101           Device identifier (from sip.conf in this
> > case)
> > > >
> > > >All of the Technology values are normal asterisk values except
for
> > APP,
> > > >which is an application (like Voicemail or MOH or MeetMe) and
PSTN,
> > > >which is a number outside of the Asterisk inside dial plan.
> > > >
> > > >I hope this helps.  Remember that for PSTN and APP values, the
> > bracketed
> > > >Extension number and the DeviceID need to be the same.
> > > >
> > > >Regards,
> > > >
> > > >Steve
> > > >
> > > > > -----Original Message-----
> > > > > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-
> > > > > admin at lists.digium.com] On Behalf Of zoa
> > > > > Sent: Friday, November 21, 2003 7:41 PM
> > > > > To: asterisk-users at lists.digium.com
> > > > > Subject: RE: [Asterisk-Users] Asterisk Call Manager for
Windows
> > 0.0.1
> > > > > (Alpha)
> > > > >
> > > > > Could you give me some explanation on how to use the
configuration
> > > >file ?
> > > > >
> > > > > I always get INVALID channels if i click on the red icon next
to
> > my
> > > >name.
> > > > > (Maybe i should use a numeric context or use numeric user
names?)
> > > > >
> > > > > Tool looks great, this will be a very cool asterisk addition.
> > > > >
> > > > > zoa.
> > > > >
> > > > > At 16:43 21/11/2003 -0600, you wrote:
> > > > > >I think the script host gets installed with Windows explorer.
If
> > you
> > > > > >don't have it, you can use the DLL in the dlls download:
> > > > > >
> > > > > >http://www.sokol-associates.com/Downloads/Dlls.zip
> > > > > >
> > > > > >Hope that helps.
> > > > > >
> > > > > >Thanks,
> > > > > >
> > > > > >Steve
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: asterisk-users-admin at lists.digium.com
> > > >[mailto:asterisk-users-
> > > > > > > admin at lists.digium.com] On Behalf Of Walker Haddock
> > > > > > > Sent: Friday, November 21, 2003 4:21 PM
> > > > > > > To: asterisk-users at lists.digium.com
> > > > > > > Subject: Re: [Asterisk-Users] Asterisk Call Manager for
> > Windows
> > > >0.0.1
> > > > > > > (Alpha)
> > > > > > >
> > > > > > > > here:
> > > > > > > >     http://www.sokol-associates.com/Downloads/AstMgr.zip
> > > > > > > >
> > > > > > > > It's written in VB6 (yes - barf, gag, whatever).  The
only
> > thing
> > > > > > > > required beyond the integral VB6 controls is the Windows
> > > >Scripting
> > > > > > > > Runtime which most PCs should have.  I will work on an
> > > >installable
> > > > > > > > version soon.  I may also port it to something more
> > > >cross-platform.
> > > > > > > > Please bear with me as I am just learning
> > Gnome/GTK/X-windows.
> > > > > > > Steve, how do you know if the Windows Scripting Runtime is
> > > >installed
> > > > > >in
> > > > > > > Windows XP Pro?
> > > > > > > Where do you get it from and how should it be installed?
> > > > > > >
> > > > > > > Thanks, Walker
> > > > > > >
> > > > > > > --
> > > > > > > ********   DataCrest, Inc. -- Technically Superior
> > > > > >******************
> > > > > > > Walker Haddock
http://www.datacrest.com
> > > > > > > DataCrest, Inc.                    e-mail:
wh at datacrest.com
> > > > > > > 1634A Montgomery Hwy.    phone:  1-888-941-3282,
> > 1-205-335-8589
> > > > > > > Birmingham, AL 35216                  fax:  1-205-823-7838
> > > > > > >
> > > > >
> > > >
> >
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