-Original Message-
From: Lance Arbuckle [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 14, 2004 6:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Parking extension not working
Sean Garland wrote:
Yes, I have the include = parkedcalls in the default context which is
where my
I have the Digium usb FXS device and an analog phone attached. How do I
transfer calls?
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt. Shasta, CA 96067
Phone: (530)926-1489
FAX: (530)926-6296
[EMAIL PROTECTED]
http://www.siskiyoutech.com
and answer the incoming call. Is that possible?
Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt. Shasta, CA 96067
Phone: (530)926-1489
FAX: (530)926-6296
[EMAIL PROTECTED]
http://www.siskiyoutech.com
to do with the T instead of t?
Thanks for you patience and help through this...
Sean
-Original Message-
From: Lance Arbuckle [mailto:[EMAIL PROTECTED]
Sent: Friday, January 16, 2004 1:28 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Parking extension not working
Sean Garland
phone, it fails because it hasn't dialed a 9 first. I
assume also that if there was a 9, it would still fail because SBC
hasn't figured out how to automatically add the 1 (like my cell phone
does).
Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt
? Thanks
Sean Garland
[EMAIL PROTECTED]
http://www.siskiyoutech.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
How would one implement a direct mailbox transfer using the macros?
What I want to do is have the person who answers the call to be able to
transfer the call directly into a persons unavailable mailbox. Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
Thanks guys I will try that in the morning...
Sean
-Original Message-
From: John Fraizer [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 12, 2004 5:33 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Direct mailbox transfer
Sean Garland wrote:
How would one implement
Occasionally I get calls that register asterisk on the caller id, which
I am assuming means that the caller id info is not there. Is there a
way to have those calls route through IVR so I don't have to deal with
them? Typically they are sales calls.
Thanks
Sean Garland
I am testing my spam filter to see if it is still catching all of the
mailing list stuff. Thanks.
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt. Shasta, CA 96067
Phone: (530)926-1489
FAX: (530)926-6296
[EMAIL PROTECTED
silencethreshold=128
maxlogins=3
append=yes
[default]
100 = 1234,Sean Garland,[EMAIL PROTECTED]
101 = 1234,Jason Madden,[EMAIL PROTECTED]
102 = 1234,Melinda Garland,[EMAIL PROTECTED]
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
___
Asterisk
to be
able to route to the internet to send out mail, so this can't be a
private subnet only pbx.
-Bryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Garland
Sent: Friday, June 11, 2004 3:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicemail
In almost all my calls now, I am getting beeps and loud and soft parts
of a conversation. It is getting very irritating. Has anyone had this
happen? How do I get rid of it?
Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt. Shasta, CA 96067
at random times. The beeps can also be heard on messages which leads me to believe its the x100p cards or something For those of you that have a fully functional system with polycom ip phones, what were the settings that worked the best to cancel the beeps and echo? Thanks a ton
Sean Garland
I bought mine from neutronexpress.com. Paid a little more then the
voipsupply.com, cuz I didn't know about them... Good to see another
place to purchase the phones... Got the firware from a poster on this
list...
Thanks
Sean Garland
Siskiyou Technology Consultants
-Original Message
I would like to setup my fax extension through freepbx to NOT have to
dial 9. I will never dial internal numbers, so all I want it to do is
pass the digits to the trunk. Is that possible with freepbx and if so,
how is it accomplished?
Thanks in advance
Sean Garland, V.P.
Siskiyou Technology
I have read on this list that the config files might be available to
make them work on Asterisk? If that is so, could someone please email
them to me? We have the Polycom Soundpoint IP 500 phones. Thanks a
bunch, my goal is to make this phone and asterisk my business system.
Thanks
Sean
very cool, and will do what I want, but it took three weeks to get my cards and the people at digium won't email to save their lives. Anyway, please help with the card problem as I feel that I am out another week and this was supposed to be running last week
Thanks
Sean Garland
Hey,
I am currently working on a Polycom 500 phone Asterisk solution, and
the key is definitely to use the xml config files that Matt spoke of.
That combined with an FTP server (setup like the sip docs say) work very
well in getting the phone to do what you want. It then becomes getting
the
to put all extensions in a single group and ring the group?
All kinda is the same question. But thanks for the answer anyway
Sean Garland
funky results..
Moral of the story is to always use two wire phone cords with the x100p fxo cards. Problem solved, and I was able to continue my development.
Thanks
Sean Garland
have read the handbook and countless searches through
wiki and Google, but cannot find practical examples of multi-line use
with asterisk.
Thanks a ton. I have been testing asterisk and on the mailing list for
about a month now... I would be happy to send all my config files for
perusal.
Sean
closer.
Thanks
Sean Garland
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt. Shasta, CA 96067
Phone: (530)926-1489
FAX: (530)926-6296
[EMAIL PROTECTED]
you to all.
Sean Garland
Siskiyou Technology Consultants
-Original Message-
From: Nicholas Comanos [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 04, 2004 8:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Multi-line help
Could you explain in a little more detail about what
Title: This is a test
It appears that my replies aren't getting to the list. Just testing to see what is going on
Sean
I have soundpoing ip 500 phones and the first few seconds of every call
has echo, which then goes away. Is there a way to have the echo cancel
on at the beginning? It seems like it is testing at the beginning but
it would be nice if I could have it start closer
Thanks
Sean Garland
Siskiyou
, with two lines,
and two Polycom phones. I hope to post my entire experience on the wiki
or something when I am satisfied that I am far enough along... Great
product, but the docs need some work. I would be interested in helping
with that when the time comes.
Sean Garland
Siskiyou Technology
you have to have at least one for access to the asterisk system, but what is the point of the multiple registrations?
Thank you to all.
Sean Garland
Siskiyou Technology Consultants
-Original Message-
From: Nicholas Comanos [mailto:[EMAIL PROTECTED]]
Sent: Sunday, January 04, 2004
Thank you all for your responses. Since I was a phone installer
(previous life) and installed Lucent Partner and Merlin systems, I was
on the key system mode of thinking. On the Polycom phones each line
button is a registration, so I wonder how I could program a SIP
registration to speed dial a
Thanks, the phones that I have Polycom Soundpoint IP 500's. In the specific config
file for the phone itself, there are some lines that have to do with MWI and there are
three settings to set. Here is the section of the manual for the phone
msg.mwi.x.subscribe ASCII encoded string
accordingly. We are using a
Mediatrix SIP gateway now - no echo at all.
Christian
On Monday 05 January 2004 13:28, Sean Garland wrote:
I have soundpoing ip 500 phones and the first few seconds of every
call has echo, which then goes away. Is there a way to have the echo
cancel on at the beginning
. ParkedCall(705)
[res_parking]
Sean Garland
Siskiyou Technology Consultants
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
I am having trouble with call parking I am basically
using the stock sample files, but extension 700 doesnt show up in my
dialplan. When I transfer a call to 700, I get the fast busy like there is
extension 700
HELP!
Sean Garland
it. This was essentially due to the 00 in
the 700, changing it to 701 eliminates the problem completely.
Hope it helps...
Girish
From: Sean Garland [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Parking extension not working
Date: Tue, 13 Jan 2004 16:07:56 -0800
I have
: Wednesday, January 14, 2004 10:31 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Parking extension not working
Sean Garland wrote:
I have just set the parking extension at 701 and then the range is
702-710 and still I cannot transfer to 701. Show Dialplan doesn't
show an extension
]
Subject: Re: [Asterisk-Users] Parking extension not working
On Wed, Jan 14, 2004 at 10:36:54AM -0700, Jared Smith said:
On Wed, 2004-01-14 at 09:15, Sean Garland wrote:
I have just set the parking extension at 701 and then the range is
702-710 and still I cannot transfer to 701. Show Dialplan
I was
running asterisk 1.0 and amp and tried to update tonight. Now I cannot load
any zaptel drivers, I get the message module wctdm not found. Im
running it on Mandrake 10.1 (2.6 kernel).
HELP!!!
Thanks guys
Sean Garland
Asterisk
1.2.4
FreePBX
2.0.1
I am running
a TDM400 card with 3 FXO and 1 FXS cards. During most all calls, there are
random beeps in the background. The other party cannot hear this (I
believe since they haven't said anything). This will happen when people
leave voicemail, and also on
on the list, and never get a response...
Thank you.
Sean Garland
Mount Shasta, CA
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
is * 1.2.4 with freepbx, the other was
* 1.0 with nothing), and different digium hardware. The only thing that
was the same is the Polycom phones, and SBC as a provider for the POTS
lines...
HELP!
Thanks
Sean Garland
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Friday, April 07, 2006 12:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Beeps and noises during calls
On Friday 07 April 2006 15:03, Sean Garland wrote:
The beeps are not DTMF tones
. To anyone experiencing issues like this make sure you check
irq sharing. I have been dealing with this for quite some time, and
getting the tdm card on its own irq, it is now working correctly and
quietly on an ASUS mobo.
Thanks again guys!!!
Sean Garland
Siskiyou Technology Consultants
I need to put a w in the
dialing prefix, but it says it isnt valid. If I manually modify the
extension file, it then affects all calls made over any trunk. Any ideas?
Sean
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database:
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] freepbx
dialing prefix
Submit a bug report to the FreePBX team?
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Sean Garland
Sent: Wednesday, April 12, 2006
8:46 PM
To:
asterisk
So how do you get a Polycom phone to work with * over NAT? I can't seem to get
it to work. If I forward ports, I can get one-way audio, but that’s it.
Looking at a packet capture, it appears that my phone is trying to send data to
the internal address of the * server, which is of course, not
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Tuesday, April 18, 2006 7:31 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Phones that work well through NAT
On Tuesday 18 April 2006 09:57, Sean Garland wrote:
So how do you get
the system try and contact a mail server.
HELP!!! Thank you all in advance.
Sean Garland
Siskiyou Technology Consultants
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
ram, is that enough? I am hoping that its my machine and upgrading will solve it. But if not, then I have a real problem.
HELP!
Thanks
Sean Garland
Siskiyou Technology Consultants s
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
48 matches
Mail list logo