, 3/24/09, Ken Williams k...@intermountainelectronics.com wrote:
I recall having similar issues early
in Asterisk 1.4...but currently running 1.4.17 and BLF works
great with a phone expiration of 15 minutes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Our work around is to lower the registration expiration on the phones.
Under account settings in the web interface on the phones, we reduced
the Register Expiration from 60 minutes to 15.
This means the phones re-register every 15 minutes...and when they
register the BLF updates. Now when
don't have 1.6 yet so I can't see if BLF behaves the same way.
--- On Tue, 3/24/09, Jon Pounder j...@inline.net wrote:
I see much the same except I think if you investigate
further, the light
will be green whether the phone ever registered or not.
--- On Tue, 3/24/09, Ken Williams k
We've had an issue since we went live nearly two years ago on Asterisk
where people complain about not being able to talk while someone else is
talking. I had assumed for a very long time this was because of the
phones we went live with (Grandstream GXP-2000's) and for the longest
time I believed
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Full Duplex
On Mon, 17 Nov 2008, Ken Williams wrote:
We've had an issue since we went live nearly two years ago on Asterisk
where people complain about not being able to talk while someone else
is
talking. I had
AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Full Duplex
Ken Williams wrote:
We've had an issue since we went live nearly two years ago on Asterisk
where people complain about not being able to talk while someone else
is talking. I had assumed
We're looking at using Global Crossing for our WAN infrastructure that's
spread across 9 states. We're hoping to gain some stability and one
point of contact for these sites, as our current infrastructure is
pathetic for VoIP.
I have a couple of questions.
1. Has anyone on this list
We're entertaining moving our intranet to Hughes satelite for our remote
locations. I'm curious if anyone with Asterisk servers has used satellite, and
if so, is the latency an issue. My understanding is that you immediately
introduce 250ms latency for travel time up and back down, however it
When I said same config I meant same with minor differences of account
information :D
[103]
type=friend
secret=1234
dial=SIP/103
callerid=Video103
allowsubscribe=yes
host=dynamic
context=from-internal
insecure=port,invite
[104]
type=friend
secret=1234
dial=SIP/104
callerid=Video104
I have 4 asterisk servers. They all have local phones on their local
network they manage for SIP based conversations. We then have IAX
between them all for inter-asterisk connections.
This setup has worked well for nearly 2 years now, minor problems here
and there but overall very nice.
I looked at quite a few options over the course of somewhere around 9
months.
We ended up going with Polycom VSX7000 series units. These units are
really designed to use H.323, but they have a SIP option that works
almost just as well. The only thing I seem to be missing that I've
noticed is
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No CallerID Transfer Problem
Try removing the quotes from the Caller*ID info.
Steve Davies wrote:
2008/4/24 Ken Williams [EMAIL PROTECTED]:
Came upon a problem today that I thought I'd see if it's by design,
if I'm
Came upon a problem today that I thought I'd see if it's by design, if
I'm missing an option somewhere, or if my fix is the way to fix it.
We setup a remote location with a server, same as we've done with
others, but for some reason when they would transfer an outside call
anywhere it would
Anyone have any experience tying the Polycom VSX 7000e Asterisk
together? It says it supports standards based SIP servers but thought
I'd see if anyone had real world experience.
Thanks,
Ken
___
-- Bandwidth and Colocation Provided by
The vast majority of what I've done with Asterisk has been with the
Grandstream GXP-2000's. These phones work great for us for everything
*except* speaker quality is quite poor and appears to be half-duplex.
So now that we've bought and are using 40 GXP-2000's we're doing some
testing on other
Having just gotten into this today, here's what I got for Grandstream,
Linksys Polycom phones to all work:
exten = _7XX,1,SIPAddHeader(Call-Info: sip:\;answer-after=0)
;this works for Linksys Grandstream
exten = _7XX,n,SIPAddHeader(Alert-Info: Ring Answer) ;this
works for
Following the recommendations here I've ordered a couple different
Polycom Aastra phones to play with speaker phones.
One of our next big projects is Video. I know Grandstream has a video
phone, has anyone used it. Anyone have any other recommendations?
Thanks,
Ken
(Who awaits the vmukti.com
I'm looking for recommendations on speakerphones for a conference room.
We're using Grandstream GXP-2000 which we've been very happy with on all
accounts, except the speaker phone. Speaker phones on these units are
extremely bad, picking up any and all background as well as having
full-duplex
PROTECTED] On Behalf Of Ken
Williams
Sent: Tuesday, October 02, 2007 10:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Zaptel slow dial out - TDM400P
Below is a copy of my log, zapata.conf extensions.conf that relate to
the ZAP lines. Basically when we
, neither made a difference.
So, before I go the bug route I'd like someone to just verify my
configuration files make sure I'm not doing something stupid.
SIP.CONF:
[general]
callerid=Unknown Caller
disallow=all
allow=ulaw
allow=gsm
[717]
type=friend
dial=SIP/717
callerid=Ken Williams 717
[EMAIL
Just a quick thanks for all being here. I started to type up a message
and realized my problem, so instead I'm saying thanks for all the good
information you all pass through my mailbox every day and giving me a
place to realize my error before I even ask the question.
Below is a copy of my log, zapata.conf extensions.conf that relate to
the ZAP lines. Basically when we dial out it takes on 10-12 seconds
before the ZAP line actaully picks up. I'm hoping to find out what the
cause is for this as it's causing user grief with extremely long connect
times, and I
I think you misunderstood the question Stephen. I believe he's saying he's got
extension.conf sub_extension.conf that's included. You can do a dialplan
reload to reload both files, but for whatever reason he only wants to load the
sub_extension.conf. Both are related to the same module, one
I used to run Asterisk 1.4.4 but had to revert back to 1.2.13 to
minimize a bug we were coming across. 1.4.5 looked promising, but the
hints are broken and making it so I'll likely have to go back to 1.2.13
until I get the hints fixed. I'm using Grandstream phones hints on
the parked
Add a line to your sip.conf:
[general]
callerid = Unknown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Erik
Wartusch
Sent: Thursday, May 31, 2007 6:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] 'asterisk' shown on display
Hi,
I've got an open bug regarding version 1.4. When I tried to find a
'stable' version I tried 1.2.18, it had the same problem. I then tried
1.2.13 and the server hasn't had an issue since.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Is there anyway of storing an incoming calls CallerID on a parked call
and having it restored when someone picks up the parked call?
I've tried storing the CID as a global variable and restoring it in my
dialplan, and while NoOp shows it working, the phone ignores it and uses
the parking lot
I fought this for a bit when I found if the file Master.csv didn't exist, it
wouldn't create it on it's own. I created an empty csv file, CDR started
writing.
Ken
From: [EMAIL PROTECTED] on behalf of Khaled Chehab
Sent: Thu 5/17/2007 10:50 AM
To: [EMAIL
It's funny Robert would come looking for this tonight, as I've been spending a
fair amount of time trying to track this down today. I then went to the source
and found what Andreas had found below.
However, I'm not a real programmer, but just a hack of a hackI tried to
make my own
http://www.pkts.ca/gsutil.shtml
I setup a script to reboot all phones using gsutil -bn 10.x.x.x 10.x.x.x and so
on for each phones address. Takes about 30 seconds to reboot 30 phones here.
Ken
From: [EMAIL PROTECTED] on behalf of Zeeshan Zakaria
Sent: Wed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E
Johansson
Sent: Thursday, May 10, 2007 12:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Problems continue...
9 maj 2007 kl. 18.14 skrev Ken Williams:
SIP
it people are ok to that,
try integrating freepbx asterisk so you know what the sip configs
should look like when things are all well.
Things might stop working if there is a bug or change in configs.
--
Deepak
Ken Williams [EMAIL PROTECTED] wrote:
I mean that SIP phones cannot answer
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Problems continue...
9 maj 2007 kl. 18.14 skrev Ken Williams:
SIP channel hang ups are progressively getting worse and I'm really
grasping at straws here trying to find out what the cause is. The
problem start
SIP channel hang ups are progressively getting worse and I'm really
grasping at straws here trying to find out what the cause is. The
problem start, once a week or so the SIP phones couldn't communicate
with the server, though there was no error message on the server and
everything appeared fine
That was in my list of things I've done, but failed to mention :). I
never have used DNS on this box, but for verification I removed DNS
servers and verified all addresses were IP's (which they were). There
is no DNS active on this box at all. There's also no freepbx, just
straight Asterisk.
: [asterisk-users] SIP Problems continue...
whats the asterisk version your using?
On 5/10/07, Ken Williams [EMAIL PROTECTED] wrote:
SIP channel hang ups are progressively getting worse and I'm
really grasping at straws here trying to find out what the cause is.
The problem start, once
I mean that SIP phones cannot answer incoming calls or make outgoing
calls. When a call comes in on ZAP, it actually rings all the phones
like normal, but when you try to answer no one is there. In addition,
when you try to dial out you eventually get a message on the phones
saying unable to
I posted about this problem last week and thought it was a combination
of SIP/ZAP causing issues in Asterisk. Since then I've realized it's
only the SIP channel that's hanging. When this happens a call can still
come in and hit the IVR, but no one can connect to the server from a SIP
client.
: Saturday, April 21, 2007 12:52 AM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Asterisk stops responding to SIP/ZAP
From: Ken Williams [EMAIL PROTECTED]
Date: Fri, 20 Apr 2007 07:27:05 -0600
About once a week or so my Asterisk box stops responding to all phones.
I can pull
About once a week or so my Asterisk box stops responding to all phones.
I can pull up the console, do whatever I want at the CLI but the only
way to get things working again is to restart Asterisk altogether.
I finally cranked verbose debugging way up (and watched my log files
go from 1mb/day
I've deployed about 40 GXP's and also haven't had the issues some are
reporting. The one issue is if I have to restart the server for any
reason I have to reboot all the phones for the BLF to light up properly.
This is easily accomplished by a script on the server, allows me to
reboot all the
Somewhere I found a link to gsutil utility that someone wrote. A quick
google looks like
http://freshmeat.net/projects/gsutil/?branch_id=59227release_id=219046
is probably the best place to get it.
This utility actually has quite a few nifty options, here's a copy of
the help:
Version 3.0 of
We're using Grandstream GXP-2000 with programmed buttons to the first 5
parking lot extensions. When a call is parked, whichever parking lot
extension it's parked on lights up red. We've never used the announce
part and I'm wondering if there's an option I can't seem to find to
disable the
?
On 3/27/07, Ken Williams [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
We're using Grandstream GXP-2000 with programmed buttons to the
first 5 parking lot extensions. When a call is parked, whichever
parking lot extension it's parked on lights up red. We've never used
the announce
?
On 3/27/07, Ken Williams [EMAIL PROTECTED] wrote:
In the basic settings, I setup the Multi-Purpose Key to use
Asterisk BLF and assigned it the parking lot extension (201 in our case,
701 by default iirc). I then added hints in the extensions.conf for the
parking lot extensions
]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Sent: Tuesday, March 27, 2007 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Park No Announce?
Paul wrote:
Ken Williams wrote:
We're using Grandstream GXP-2000 with programmed buttons
] on behalf of Ken Williams
Sent: Tue 3/27/2007 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Park No Announce?
The problem isn't on the outside phone, it's on the inside. An outside
caller already gets MOH immediately, the problem comes in waiting
It's been up since early morning for me. Just refreshed - still up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent: Wednesday, March 14, 2007 12:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Try
exten = s,1,GotoIf($[${ARG1:0:5}=220408]?2:3)
This looks at the first 5 digits of ARG1.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ricardo
Carvalho
Sent: Tuesday, March 13, 2007 12:39 PM
To: Asterisk Users Mailing List - Non-Commercial
I don't believe this will work. He wants it to goto if EXTEN =
220408235 or 220408743 or any other digits for the last 3 of the
extension block 220408xxx. When Asterisk processes both his and your
line it's going to look to see if the EXTEN is exactly 220408XXX, which
of course it will never be.
In case it hasn't been posted before, here's instructions to get the
correct time to show up on your Grandstream GXP-2000's:
1. Login to phone
2. Go to Basic Settings tab
3. Change Daylight Savings Time to yes
4. Change Optional Rule to 3,2,7,2,0;11,1,7,2,0;60 (this means change
clocks the
Every few seconds I get the following message:
== Parsing '/etc/asterisk/manager.conf': Found
== Connect attempt from '127.0.0.1' unable to authenticate
I'm trying to track down where it's coming from.
I've used TCPDUMP NGREP to monitor 127.0.0.1, no data's flowing.
I've tried
your op_server.cfg file (/var/www/html/panel/, I think). Look for
the manager_user and manager_secret parameters, and make sure they match
an entry in /etc/asterisk/manager.conf.
Alex
On 3/6/07, Ken Williams [EMAIL PROTECTED] wrote:
Every few seconds I get the following message
1. We just dial the extension directly and have speed dials setup for
the first 6 parked positions. We don't use *8 at all.
2. Change the config on the phones under Account to Send DTMF via RTP
(RFC2833)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2 is one of
the easiest configs to put together. Works extremely well and requires
opening a single port on each NAT.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Thursday,
I am currently running Zaptel 1.4 with Asterisk 1.2.
From: [EMAIL PROTECTED] on behalf of Trevor Peirce
Sent: Fri 1/12/2007 4:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] zaptel asterisk versions (was Echo...)
wrapped it up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: Friday, January 12, 2007 9:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Echo...
On Jan 11, 2007, at 8:53 PM, Ken
I've spent all day today trying to fix an echo problem and I've made no
ground whatsoever.
I have Digium TDM400 with 3 FXO 1 FXS. I've tried this computer at
two completely different sites with different phone providers. I've
tried compiling installing different versions of Zaptel (currently
I tried to be thorough, but of course left fxotune out. I did try fxotune, it
resulted in something like 9,0,0,0,0,0,0,0 for each fxo (I'm not at work now, I
can post the results if it'll help).
Ken Williams wrote:
I've spent all day today trying to fix an echo problem and I've made
I've never understood why people would think it's a PSTN issue. I'm sure
99.99% of Asterisk users are using Asterisk on lines they've had regular
phone lines hooked up to before moving to Asterisk. In my case I've also tried
having the server plugged in to two systems that are 200 miles
You need the kernel source installed to compile Zaptel.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
Torrez
Sent: Tuesday, December 26, 2006 4:45 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] I cant install zaptel
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] BLF on GXP2000
Rebooted the phone...No luck
On 12/19/06, Ken Williams [EMAIL PROTECTED] wrote:
One thing I've noticed, is any time I make changes to Asterisk I
have to reboot the phones to keep
=-
[EMAIL PROTECTED]: SIP/101
State:Idle Watchers 1
- 1 hints registered
On 12/18/06, Ken Williams [EMAIL PROTECTED] wrote:
Here's what I have, it's to early for me to think so hopefully
looking at mine helps :D
Here's what I have, it's to early for me to think so hopefully looking
at mine helps :D
extensions.conf:
[ext-local]
exten = 701,1,Macro(exten-vm,701,701)
exten = 701,n,Hangup
exten = 701,hint,SIP/701
sip.conf:
[701]
type=friend
secret=1234
record_out=Adhoc
record_in=Adhoc
qualify=yes
I was able to set a program to speed dial the park extension. Then a user just
hits TNFR followed by the line I've programmed to speed dial park.
If you get the HOLD button to do this, I'd love to hear how :).
From: Steve Sobol
Sent: Fri 12/1/2006 5:15 PM
To: Asterisk Users Mailing List
I'm trying to use a
simple page function. It starts a MeetMe conference with the devices I've
listed, but the devices hang up after 3-5 seconds. After doing some
research I found this was a problem, and I needed to remove a (5) from
app_page.c
Well, my app_page.c
didn't have the (5). I
BAH!
My Makefile in the apps folder was missing
app_page.c. I added it, recompiled, page is working
properly.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
WilliamsSent: Wednesday, November 15, 2006 10:33 AMTo:
Asterisk Users Mailing List - Non-Commercial
We use intercom 100%
inter-office. To get FreePBX to do this with Grandstreams by default
without having to create intercom or paging groups, just change the following
line (line #58) in your extensions.conf from:
exten = s,10,Dial(${ds})
; dialparties will set the priority to 10 if $ds is
I've installed Grandstream GPX-2000 phones and have successfully enabled
one of my buttons to use Asterisk BLF for an extension. I can tell when
this extension is available, is being rung, or is on the line.
I'd like to do the same for my Zaptel channels, to be able to see when a
line is
I've installed Grandstream GPX-2000 phones and have successfully enabled
one of my buttons to use Asterisk BLF for an extension. I can tell when
this extension is available, is being rung, or is on the line.
I'd like to do the same for my Zaptel channels, to be able to see when a
line is
doesn't like them. When I do show
hints in CLI it is registering both 102 702 properly.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
Williams
Sent: Wednesday, November 15, 2006 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial
A day of banging my head against a wall and spamming this list is about
done
I've got everything working beautifully, and I'm ready to go full out
and implement all across the board...save for one stupid little thing.
We have 6 phone lines and I'd like the GXP-2000 to show the status of
After about one
weeks time I've gone from no VoIP to a completely configured system for two of
our offices to be able to page/communicate interoffice as well as handle
existing PSTN communications (okay, waiting onf hardware for the PSTN side and
I've likely jinxed myself now).
I was
I was planning on
using a TDM400P with 3 FXO 1 FXS, with the 1 FXS being used for a fax
machine. It now appears that Digium doesn't support this, are there other
manufacturers anyone can recommend that will support it? Has anyone used a
TDM400P in this setup and had it work without much
supply
company (the place with a long counter and lots of electricians drinking
coffee ordering their parts.).
Andrew
On 10/31/06, Ken Williams [EMAIL PROTECTED] wrote:
I knew I should've waited til tomorrow to send the e-mail so I could
have a nights thought on the subject.
That being said
I've been doing a lot of reading over the last few weeks on Asterisk,
and will be implementing a test system this week to play with.
I've got two questions in regards to the ideal implementation for our
company. First, has anyone written any drivers to interface with
proprietary phones?
I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject.
That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our
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