Anyone having issues
with the message indicator lights after CVS-HEAD-07/23/04-13:55:59
??
For several of my
users, our MWI lights do not turn off. Phones are Polycom IP500 and this
just started prior to my last update.
Should I update to a
newer version? I pulled this from the CVS last
Interestinging
From my voicemail.conf,
my context where I
define my mailboxesin this file is
[sip]
In the sip.conf I have [EMAIL PROTECTED]
Changed that to [EMAIL PROTECTED]
and it seems to work better now.
Thanks!
Wiley
From: Paul Rodan [mailto:[EMAIL PROTECTED]
Sent:
From my voicemail.conf,
my context where I define my mailboxesin this file is
[sip]
From: Paul Rodan [mailto:[EMAIL PROTECTED]
Sent: Monday, November 08, 2004 2:49 PMTo: 'Asterisk Users
Mailing List - Non-Commercial Discussion'Subject: RE:
[Asterisk-Users] MWI Doesn't Turn Off
What is
Well, the phone automatically does call waiting on each line you
register so you will be able to get a call on each line.
You could always do this
In Asterisk
Setup extensions 100, 101, 102, 103, 104, 105, 106
Set your dial plan to ring 100 on all incoming calls.
Set 100 to roll through
Any documentation on how to do this
anywhere?
W
From: Brian C. Fertig
[mailto:[EMAIL PROTECTED] Sent: Thursday, October 14, 2004
12:52 PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [Asterisk-Users] Running Asterisk on Linksys
Router
ahh my bad..
Didnt know
I have Polycoms and I sometimes have echo. However, it is always on an
incoming call and always a matter of echo training.
Have you already worked through all the settings for ech in *?
Are you adjusting rx and tx gains?
Do you have echo cancellation set to high?
What are the rest of the
99.9% sure not sound card is required for MOH.
I don't think you want the latest version of MPG123. Think you want
mpg123 0.59r only not s-r4
Make sure to copy all mp3 files (if over FTP) using the binary transfer
method only.
W
-Original Message-
From: Andy Reinke [mailto:[EMAIL
Go here..
http://www.voip-info.org
search on IVR
Everything is driven out of
extensions.conf
Regards,
Wiley
From: Jose J. Avalis [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 23, 2004 12:42 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] new user
>From Canada
Hi all,
We are
Search here... http://www.voip-info.org
There are alternatives...
-Original Message-
From: Leah Newmark [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 23, 2004 1:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Alternate MP3 Player
Hi! I am currently working on setting up
Do you have a price range?
I use Polycom IP500s and the speaker phone is awesome. It picks up
speakers in the room very well at 5-6 feet.
Polycom has always made an exceptional speaker phone even on plain ole
phones.
Their implementation on the IP phones is excellent so they are my
preference.
I
The free phones I have heard of are soft phones...
X-Lite is excellent...
Cheap phones to test with, Grandstream is cheap but like the man said,
you get what you pay for.
eBay is a good source for cheap phones to test with but cheap is
relative
I consider cheap as sub $100.
You can pick
Here is the best starting point. It is all driven out
of extentions.conf
http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20ivr%20menu
Search for IVR and you will find good
info...
Cheers,
W
From: Luis Czop [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 21, 2004
I use my XLite softphone from my Win XP box over VPN to my
Cisco PIX with no issues so this can be done.
How that works for an ISA box is unknown to me. I
dumped ISA several years ago do to it's (IMHO) unpredictability and low
performance.
Are you using the built in VPN of WinXP or an ISA
See my post of a few moments ago and you have hit on the exact reason I
will not use Cisco beyond my firewall (a purchase you will never regret
if you need a good firewall). Cisco makes arguably some of (if not
totally) the best equipment out there. I just have one problem.
Their licensing model
.
--On Tuesday, September 21, 2004 15:25 -0700 Wiley E. Siler
[EMAIL PROTECTED] wrote:
See my post of a few moments ago and you have hit on the exact reason
I will not use Cisco beyond my firewall (a purchase you will never
regret if you need a good firewall). Cisco makes arguably some
: Michael Loftis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 21, 2004 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Cisco 7905G
--On Tuesday, September 21, 2004 15:58 -0700 Wiley E. Siler
[EMAIL PROTECTED] wrote:
A completely valid point
Bruce,
Using a POTS line local with * will get you the same net result as
having the POTS line only. You will be using VoIP internal and passing
your calls off to the * box to have it dial like a normal phone. So, no
IP packets move past the box over a POTS line. That is a pretty useful
Here you go... No extension required
From extensions.conf
;--
; VOICEMAIL ENTRY INTO SYSTEM
;--
exten = 8,1,Answer
exten = 8,2,Wait(1)
exten = 8,3,VoicemailMain(${CALLERIDNUM})
exten = 8,4,Hangup
won't
work.
Should I have the callerid set somewhere else?
Matthew
- Original Message -
From: Wiley E. Siler [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, September 20, 2004 4:14 PM
Subject: RE
Did you transfer the mp3s to the * box via FTP? If so, did you use
binary not ascii mode?
Ascii mode will mess the files up every time. Explicitly call binary
then your mget so your files don't get hosed.
Also, if your musiconhold.conf file has a good reference to where your
mp3s are located...
Felix,
You might try going to Sam's (or their website) and looking
for the motherboard manufacturer on their marketing materials. Then you
can get the specs for the motherboard from the mobo maker. I cannot
imagine anyone here will know if the PC you are reference is compliant of the
tops
I never stripped my tags and things work fine for me. I had problems at
first too with MOH. My problem was due to how I was copying over the
files. I was copy via FTP using the command line in Linux. However, if
you do not explicitly state binary as the copy method, it will copy the
files over
on a PPC, I'd be interested to hear of your
experience.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. Siler
Sent: September 11, 2004 12:36 AM
To: Asterisk Users
Does anyone know if
SIP will/is support on handheld PCs such as the iPaq or Axiom? With their
integrated 802.11b and Bluetooth it seems like a solution to provide a wireless
based sip phone for any user would be possible. Handoff between access
points might be problematic but most users I
/products/TypeLanding/0,1056,112,00.html
and there's a client available to connect to iax on voip-info.org. I
know you asked for SIP, but.. this is all that's avail i can find :)
http://www.kauss.org/Stephan/ziaxphone/
matt
Wiley E. Siler wrote:
Does anyone know if SIP will/is support on handheld
] SIP on Handhelds
As far as I can tell, both SJphone and X-Lite offer PocketPC versions.
--On Friday, September 10, 2004 4:25 PM -0700 Wiley E. Siler
[EMAIL PROTECTED] wrote:
I think the Bluetooth requirement may be where that hangs up. I want
to be able to setup an handsfree headset too
Also, is the new SIP and bootrom release available for download
somewhere?
Thanks,
Wiley
-Original Message-
From: Derek Listmail Acct [mailto:[EMAIL PROTECTED]
Sent: Monday, August 16, 2004 2:39 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Polycom SoundPoint IP 500/600 XML
Is there a Software
based DSS application available for Asterisk?
Thanks,
Wiley
Siler
-08-17 at 00:34, Wiley E. Siler wrote:
Also, is the new SIP and bootrom release available for download
somewhere?
Thanks,
Wiley
Don't know if these are the latest but here are some links.
First one has sip bootrom files:
http://www.freedomphones.net/polycom/files/
http
Best starter
examples
http://www.automated.it/guidetoasterisk.htm
Documentation
http://www.digium.com/index.php?menu=documentation
Asterisk will make sample files for
you... read teh doucmentation at the first link I
listed...
Regards,
Wiley
I need to find some basic configuration
Hello Francis,
My office build is the same as yours. 15 or so extensions, low traffic
100MB network, and a desire to have a phone system that uses VoIP. I
have my system working as a PBX just like you would. I use two TDM400s
for my 8 POTS lines and Polycom IP 500 phones at the desktop. I
Hello Francis,
I'll most likely use a BRI. Do you think this will help to avoid echo?
I could not say as I have never used a BRI and I am pretty new to this
too. I do know that BRI is supported from watching conversations in
this email list and reading online. People seem to use it a bit so
Hello All,
Sorry to rehash a question I am sure has shown several time
but I cannot google up the answer from the lists.
Does anyone know where I can get some royalty free, cost
free music for my music on hold?
I saw someones post several weeks ago that said that
this exists at a
://www.sounddogs.com/catsearch.asp?Type=2 Royalty
Free
Music
* FreeMusic http://hebb.mit.edu/FreeMusic/ Free Classical Music
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley E.
Siler
Sent: Saturday, August 14, 2004 7:51 PM
To: [EMAIL
Hello
All,
I have Polycom IP
500 phones which I would like to have message waiting indicators on. So
far, I have my system running well but the problem I am seeing is that MWI
doesn't seem to tell my phone that it should display a MWI state. The
light does not show when you have message
Nope. That fixed it. Thank you!
Wiley
-Original Message-
From: Robert Jackson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 2:16 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Polycom IP 500 - MWI Not Working
-Original Message-
From: Wiley E. Siler
Hello
All,
Does anyone know the
state of AstMan? I found some information and source code in the archive
but it is from November of 2003. There is mention of a lgpl release but
nothing else after. I would like to code in some of the features that were
lacking like setting this in system
THat is part of your phone configuration file, not
Asterisk. Look in cfg files for your sip phone. If should be in
macaddressphone.cfg
dialplan
dialplan.1.digitmap="9[2-9]xx|91xx|9xxx|[1-8]00x|9411|9911|90"
dialplan.1.digitmap.timeOut=""
MP3s have to use constant bitrate not variable bit rate. Look in the documentation
for mpg123.
-Original Message-
From: Jozeph Brasil [mailto:[EMAIL PROTECTED]
Sent: Saturday, July 24, 2004 5:30 AM
To: [EMAIL PROTECTED]
Subject: RES: [Asterisk-Users] Play CD!
I do that. But when
Hello
All,
I rebuilt my machine
adn there has been about 2 weeks time since my original CVS checkout. I
have seen teh changes for features.conf so that does nto worry me.
Hwoever, after moving over my saved conf files, things are not really
running. Does anyone know aht happened to teh
is this?
[outgoing]
exten = _9.,1,Dial(ZAP/g2/${EXTEN,1})
What is Zap/g2? I don't see group 2 given in zapata.conf.
John
Wiley E. Siler wrote:
Actually, I am having trouble with my X100P setup too which will
probably sow when you read through my configs. I cannot get my
referencing
.
And what is this?
[outgoing]
exten = _9.,1,Dial(ZAP/g2/${EXTEN,1})
What is Zap/g2? I don't see group 2 given in zapata.conf.
John
Wiley E. Siler wrote:
Actually, I am having trouble with my X100P setup too which will
probably sow when you read through my configs. I cannot get my
Whenreceiving
an incoming call,I get sent to my IVR just fine. My Playback event
plays back my test file and the itis suppose to Hangup. The Hangup app
fires off and I see the console say it hungup the line. However, I cannot
receive anymore calls after that. When I run 'zap show channel 1'
Hello
All,
I have a system up
and running that will be used as a PBX lcaolly with SIP phones. Because I
am dumping all my calls into my X100Ps and have a very small number of clients
(15), I woudl like to set all my call quality variables to their highest
levels. I ahve a 100 meg network
Install the kernel-source RPM off of the RH9 CD.
-Seth
On Tue, 2004-07-20 at 20:28, Wiley E. Siler wrote:
The error I receive when I run make
Thanks,
Wiley
-Original Message-
From: Wiley E. Siler
Sent: Tuesday, July 20, 2004 4:12 PM
To: [EMAIL PROTECTED]
Subject: RE
. (Asterisk discerns the mailbox from
the extension of the phone) It's one touch (but still password
protected) It's working here and I'm sure we can get it to work at your
office.
Voicemail answers on extension 8.
Just to be sure, can I see your extensions.conf?
John
Wiley E. Siler wrote:
I
A. Icide wrote:
On 04:28 PM 7/19/2004, Wiley E. Siler wrote:
Mine does the same. Once in Message center I can choose selection
1.Message Center and then soft key Select.Then I select the
registered line that I want to check voice mail on. That is no less
than
4 key strokes just
I attempted to
install an X100P card but it was not correctly recognized by my Redhat 9
install. I had a test install running without any cards which was working
great minus the outward dialing since no cards existed. Now that I have a
card, I want to add it to the system. Do I have to
I attempt to run
make clean:make install and I get the following (cut short for
brevity).
zaptel.c: In function `zt_init':zaptel.c:6123:
warning: implicit declaration of function `register_chrdev'zaptel.c:6124:
`KERN_ERR' undeclared (first use in this function)zaptel.c:6124: parse error
When I run make I get all kinds of errors. So far I
ahve yet to get past that problem and when I look for /etc/zaptel.conf and
/etc/asterisk/zaptel.com these fiels do not exist.
W
From: Celedonio Albarran
[mailto:[EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 12:57
PMTo: [EMAIL
source directory and then make clean make install in
asterisk source directory.
-Seth
On Tue, 2004-07-20 at 13:54, Wiley E. Siler wrote:
I attempted to install an X100P card but it was not correctly
recognized by my Redhat 9 install. I had a test install running
without any cards which
. You don't have to change your version, just make install in
zaptel source directory and then make clean make install in
asterisk source directory.
-Seth
On Tue, 2004-07-20 at 13:54, Wiley E. Siler wrote:
I attempted to install an X100P card but it was not correctly
recognized by my Redhat 9
.
John
On Tue, 2004-07-20 at 11:52, Wiley E. Siler wrote:
I tried this configuration and it still does not work for me. In
fact, now I cannot dial in using the menu system of the message
center. Here is how I have now mine configured and what I get...
msg msg.bypassInstantMessage=1
Wiley E. Siler wrote:
I have a solution that allows me to assign a soft key with no
problems.
However, it seems like a waste the the hard button labeled Voice Mail
is not dialing right into voice mail. Is there a known way yo do
this? I have tried everything in the manual but it doesn't
I read the administrator document repeatedly. I have not been able to
find a wiki that applied to digitmap feature at all and I have searched
repeatedly and read several of the wikis regarding Polycoms. The
administrators guide doesn't have enough context explanation to make the
use of the
Interpretation
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: Monday, July 19, 2004 11:50 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail
On Mon, 2004-07-19 at 12:42, Wiley E. Siler wrote:
Thank you!
Can you tell me more about the dial
-Users] Polycom IP 500 Voicemail
On 12:40 PM 7/19/2004, Wiley E. Siler wrote:
My Polycom is on loan as a demo and I assume it is one of the first
revision models. In fact it shows as Rev A on the back of the phone.
I have all the same buttons you listed save for the Messages button.
The 3rd from
Is this bascially setting your bandwith value = high inside of iax.conf?
Or is there another place to designate the codec?
Thanks,
Wiley
-Original Message-
From: Senad Jordanovic [mailto:[EMAIL PROTECTED]
Sent: Monday, July 19, 2004 2:11 PM
To: [EMAIL PROTECTED]
Subject: RE:
Wiley E. Siler wrote:
I read the administrator document repeatedly. I have not been able to
find a wiki that applied to digitmap feature at all and I have searched
repeatedly and read several of the wikis regarding Polycoms. The
administrators guide doesn't have enough context explanation to make
I think I saw a reference to a similar problem and it regarded IRQ
issues on the machine in question. IF there was IRQ sharing, cagey
things happened. But if the T1 card had a static IRQ, it resolved the
issue. Does your T1 card have a dedicated IRQ? I am sure someone will
be able to explain
Hello
All,
So far I have been
unable to get the hard button labeled Voice Mail to conenct to Asterisk. I
have followed all the Admin Guide instructions regarding the .cfg files and
using up.bypassInstantMessage="1" up. to no avail.
Has anyone been able to get a Polycom 500 to use the
I would comment out these lines in sip.conf
;externip=111.222.333.444
;localnet=192.168.1.0
;localmask=255.255.255.0
Then set nat=no
-Original Message-
From: Simon Chappell [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 4:49 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
I just started out too and I can tell you it is easier to start from
scratch with a good wiki then alter the demo files. Here is a wiki you
can build a good working system with...
http://www.wlug.org.nz/AsteriskSampleSetup
For your ciscos search http://asterisk.xvoip.com/index.php
Wiley
Hello
All,
I have some Polycom IP 500 phones that I would like to
have configured for direct dialing to our voice mail system.
So far I have been unable to get
the hard button labeled Voice Mail to connect to Asterisk
without first passing through the message center prompts. I have
dialing a number?
Thanks for the tips!
Wiley
-Original Message-
From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 5:56 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail
Wiley E. Siler wrote:
Hello All,
I have some Polycom IP 500
Does anyone know
where I can find a list of all the control scripts? I want to write a
standard windows tool that will allow you to pregenerate the configuration for
your Asterisk install and them press one button to have it log into your
boxand upload the scripts. Of course, I will let
I ahve been
searching to no avail for a referenc eon how to setup a part of my dial plan
that will ring certain groups of number based upon the context.
Essentually, I want to be able to designate 3 people as sales and have my IVR
handoff and ring their extensions in order. Then maybe I
,URL)
-Seth
On Sat, 2004-07-17 at 19:24, Wiley E. Siler wrote:
I ahve been searching to no avail for a referenc eon how to setup a
part of my dial plan that will ring certain groups of number based
upon the context. Essentually, I want to be able to designate 3
people as sales and have my
...
-Seth
On Sat, 2004-07-17 at 22:05, Wiley E. Siler wrote:
That could be it. What I want to do is set a group of callers and
have the event cause the phone to ring them in order. I will tie it
to my IVR portion and thus I can make sure peole in sales get calls
based on our hierarchy
Hello
All,
Thanks for all the
great info!
Is there anyone out
there using Polycom IP 500 phones with Asterisk who can advise on how to get
these phones easily configured? So far, I have ben unable to google up any
tools for them and the example config files I have do not have any
It absolutely ships with Windows 2K/XP versions. Regsvr32 will work
from any folder on a standard install.
Wiley
-Original Message-
From: Stephen R. Besch [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 1:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Updated
Steve,
For your batch, be sure to include the /s switch after the command so it
runs silent (no prompts)
Thanks,
Wiley
-Original Message-
From: Stephen R. Besch [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 1:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Updated
I think that is also an ID ten T problem.
W
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 8:46 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] [OT] The stories people tell to support.
On 15 Jul 2004 at 20:48, Dave Cotton
Do you have these values set?
externip
localnet
localmask
-Original Message-
From: Harold Workman [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 13, 2004 1:25 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problem with multiple phones behind firewall
Hi,
I am having a problem
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