Please take a look at http://www.snom.com/howto40.html. We tried to make
the upgrade procedure as smooth as possible, if you are having problems
please tell us and we will try to make it more simple. For example, if
you have a batch of phones give us an email and we will send you the
files in one
And where did you get your rate?
The 11/2004 rates from nufone show:
Taiwan 886 0.0469
Taiwan - Mobile/Special Services886 60 0.1006
Taiwan - Mobile/Special Services886 70 0.1006
Taiwan - Mobile/Special Services886 9 0.1006
Taiwan -
Just buy hardphones, problem solved...
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Raja
Chidambaram
Sent: Saturday, August 06, 2005 12:45 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Phone interface hardware
---
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 05, 2005 7:39 PM
Subject: Re: [Asterisk-Users] Is this echo problem down to IP Phone
hardware?
Kris
my customers complain that when they make a call they
hear the another side very well but the another side
hears the first side well but in low sound.what is the
ptoblem here and i have to change?
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has
On Sat, 6 Aug 2005, Angus Comber wrote:
I have a Grandstream GXP2000 with latest firmware. When I use it holding
the handpiece I don't hear any echo - neither does other end. However,
if I use it handsfree, the other end notices echo when they speak - ie
their voice is echoy. I hear
Hi all,
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success
stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4).
Please any comments?
Kumara
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Asterisk-Users@lists.digium.com
Kumara Jayaweera wrote:
Hi all,
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success
stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4).
Please any comments?
Kumara
___
Asterisk-Users mailing list
I went to run my queue_log parser so that I could send out a monthly
report to one of my customers, and I noticed that every valid call
complete action (COMPLETEAGENT, COMPLETECALLER) is followed by an
ABANDON:
Here is a complete-caller:
1123325015|1123325011.2|mainq|NONE|ENTERQUEUE||00110102102
Hi, all
I am trying to set up voicemail. I've done it to the point where I can leave
messages.
How do I retrieve them?
Actually I have few questions:
1. I want voice mail to be available at certain extension, say 100. How do I
set it up so all users can call this number and get to their
1) Create the following in your dialplan:
exten = 100,1,VoiceMailMain()
2) Set their password to 1234. They can change it in the voicemail menu.
3) See: Getting MWI on Polycom Phones to work with Asterisk
http://www.voip-info.org/wiki-Getting+MWI+on+Polycom+Phones+to+work+with+Ast
erisk
I
Thanks a lot,
Will try tomorrow.
Rudolf
- Original Message -
From: Cullin J. Wible [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Saturday, August 06, 2005 10:08 PM
Subject: RE: [Asterisk-Users] Voicemail --
A few comments:
1) We are using quite a few SoundPoint 300 and 501's with no problem.
2) Your intermittent ring issues sounds like what we saw when we tested the
phones through NAT (which doesn't really work at all despite what the
documentation says).
3) We also upgraded to the latest boot
It sounds like you need to adjust your txgain or possibly rxgain.
MARK.
jonny hashem wrote:
my customers complain that when they make a call they
hear the another side very well but the another side
hears the first side well but in low sound.what is the
ptoblem here and i have to change?
I don't use Fedora but I do use RHEL AS 4 without any problem. Do you
have any USB conflicts?
MARK.
Kumara Jayaweera wrote:
Hi all,
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success
stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4).
Please any
Hi all,
When installing Asterisk onto Fedora Core 4, do I need to modify anything in
the system for Asterisk to work fully out? I will install:
zaptel, libpri and Asterisk in that order. Will the ztdummy driver work
without any further modifications except to the make file?
Many thanks,
On Sat, 2005-08-06 at 16:14 +0600, Madhawa Jayanath wrote:
Kumara Jayaweera wrote:
Hi all,
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success
stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4).
Please any comments?
Kumara
Without having said why you want to connect the phone through the
computers I have a hard time understanding why you want to do that. As
someone else has suggested, either buy IP phones and connect them
directly to your LAN or buy analog adapters and use your existing phones.
I don't see the
Run memtest86 from the boot menu. You may have faulty RAM. I had the
same problem installing CentOs 4...
Julian J. M.
On 8/6/05, Kumara Jayaweera [EMAIL PROTECTED] wrote:
Hi all,
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success
stories? my Intel 865 M'd+ Intel 3.0GHz
Hi,
can anybody tell me how to create an extension that starts with a *? The
expression matching works well if * is embedded in numbers but if the
extension starts with *, it is not executed but extension s instead. Is
there another way besides using a lot of if statements in the s extension?
I'm seeing this same issue. The following message will popup on the console:
-- Attempting native bridge of Zap/1-1 and Zap/74-1
At the same time my call is briefly muted, I hear a quick DTMF tone, then it
unmutes. The whole process takes about 1.5 seconds. Is there any way to stop
these
On Fri, Aug 05, 2005 at 06:30:30PM -0600, Joseph wrote:
Is there any script guru on the list that can help me.
I'm trying to start asterisk with nice -5.
Normally the command would be:
nice -5 asterisk
but asterisk start from the scrip on Gentoo as -U asterisk -G asterisk
Here is the
Any different opinions if I say I'm getting the phones for $40ea? Sure
seems like some others had good luck with these phone.
-jim
-
I can only advise against them. they are like a wallmart special. Barely
Just got in a bunch of polycom phones for use on my shiny new
asterisk box, but found 2 small issues I was wandering if someone
could help me with.
Are you using AMP or Asterisk @ Home?
First, though the phones support 2 call appearances, if I am on a
call, the second call does not ring
Can anyone tell me if the CallerID information is automatically displayed on
the LCD screen of the 301?
Can asterisk manipulate the LCD screen for the purposes of displaying
callerid?
Is this a good quality phone? Or, is the 501 worth the added expense?
I believe the only real differences
I'm using 1
BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The
analog phones with the Sipura seem to work great. Voice quality is fine on
both ends on the Sipura. I'm using the Teliax service and I use the Ulaw
codec for all phones.
However, I'm
struggling with the
The screen on the 301 is crap. I would definitely advise paying the
extra for a 501.
Jim Duda wrote:
Can anyone tell me if the CallerID information is automatically displayed on
the LCD screen of the 301?
Can asterisk manipulate the LCD screen for the purposes of displaying
callerid?
Is
I've been searching the forums and on the list to see if this has been
addressed. If it has, could someone point me to the thread to fix or at
least acknowledge it is an issue and what is causing it. Posting to the list
was last resort as I couldn't find a solution anywhere else.
Setup:
[EMAIL
On Friday 05 August 2005 21:31, Doug Lytle wrote:
exten = s,1,Dial(SIP/PHONE1,15,rt)
exten = s,2,Dial(SIP/PHONE4,15,rt)
Using 'r' flags makes baby Jesus cry. Stop doing that.
-A.
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Asterisk-Users@lists.digium.com
Jim Duda wrote:
Can anyone tell me if the CallerID information is automatically displayed on
the LCD screen of the 301?
Can asterisk manipulate the LCD screen for the purposes of displaying
callerid?
I find the 501 to have a higher resolution and better quality on the
speakers. Both
On Sat, Aug 06, 2005 at 09:02:30AM -0600, Mike Putnam wrote:
Any and all suggestions are greatly appreciated.
Provide some useful input:
* the relevant config files
* a CLI trace of the course of a problematic call, after you have set
verbosity to a resonable value. E.g: 'set verbose 3'.
On Friday 05 August 2005 21:10, Arik Funke wrote:
how can I implement a dial plan like the following:
Well you've scripted it out very clearly... now just use the dialplan
applications and make it exactly as you've said.
1. ring phones 1,2,3 monday to friday between 9:00 and 20:00; if no
Ryan Stark wrote:
I went to run my queue_log parser so that I could send out a monthly
report to one of my customers, and I noticed that every valid call
complete action (COMPLETEAGENT, COMPLETECALLER) is followed by an
ABANDON:
Here is a complete-caller:
I second that motion. I have a 500, and recently picked up a 300 and
wish I would have bought another 500 instead :-(
I felt that since I didn't need a speakerphone, the 300 would be fine.
The phone itself is smaller, and the LCD really sucks in comparison. As
for voice quality, they both are
Tim,
Can I test it as well?
Best Regards,
Newbie
On 8/2/05, Vlasis Hatzistavrou - asterisk mailing list account
[EMAIL PROTECTED] wrote:
If anyone is interested I'm (slowly) developing a GPL'd Java applet that
works as an IAX softphone.
I should have a test version out at the end of
I'm trying to set-up H.323 support under Asterisk. I built a recent CVS
release and the ooh323c code from the asterisk-addons. Everything built
and installed and the H.323 stuff loads OK when asterisk starts.
What is the easiest way to check if the H.323 code is working? I've
edited the
On Sat, 2005-08-06 at 14:21 +0100, Julian J. M. wrote:
Run memtest86 from the boot menu. You may have faulty RAM. I had the
same problem installing CentOs 4...
Julian J. M.
On 8/6/05, Kumara Jayaweera [EMAIL PROTECTED] wrote:
Hi all,
Does anyone run Asterisk on FC4? with Digium's
Hi,
I have researched more into the problem of my Asterisk set-up not answering
calls.
The following error was shown on the CLI, can anyone explain what the
problem causing Asterisk to not answer the SIP calls be?
Information: I have an Asterisk box on a home LAN, behind a D-Link
Check your zapata.conf file. Your terminal profile options section under
[channels] should be ended by adding the associated channel, i.e. channel =
1
Sample two port config:
[channels]
;
; Default language
;
language=en
; Default terminal profile FXS PORT1
;
context=internal
signalling=fxo_ks
On 8/6/2005, Frank Tarczynski [EMAIL PROTECTED] wrote:
I'm trying to set-up H.323 support under Asterisk. I built a recent
CVS release and the ooh323c code from the asterisk-addons. Everything
built and installed and the H.323 stuff loads OK when asterisk starts.
What is the easiest way to
Andrew Kohlsmith wrote:
On Friday 05 August 2005 21:31, Doug Lytle wrote:
exten = s,1,Dial(SIP/PHONE1,15,rt)
exten = s,2,Dial(SIP/PHONE4,15,rt)
Using 'r' flags makes baby Jesus cry. Stop doing that.
Excuse me?
___
Asterisk-Users
chris gamble wrote:
The second is: a lot of my phones will not ring for internal
extensions. They show up on the screen as a call ringing in, but the
phone itself wont ring. About 50% however do ring. What could cause
this?
One of the few REALLY stupid things that I've found with the Polycoms
[EMAIL PROTECTED] is believed to have said:
--
Message: 9
Date: Thu, 4 Aug 2005 07:43:51 +0200 (CEST)
From: Bastian Scholz [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] chan_capi upgrade
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
On Sat, 2005-08-06 at 11:36 -0500, Larry Shields wrote:
Check your zapata.conf file. Your terminal profile options section under
[channels] should be ended by adding the associated channel, i.e. channel =
1
Sample two port config:
[channels]
;
; Default language
;
language=en
;
If it is expensive to get a separate LAN connection for analog phone
adapters, you can get one with 2 ethernet port and 1 FXS port such
that
it can connect your PC and analog phone over a single cable to the
network. It is not difficult to find such kind of analog adapters for
around US$50 or
On Sat, 2005-08-06 at 13:02 -0400, Doug Lytle wrote:
Andrew Kohlsmith wrote:
On Friday 05 August 2005 21:31, Doug Lytle wrote:
exten = s,1,Dial(SIP/PHONE1,15,rt)
exten = s,2,Dial(SIP/PHONE4,15,rt)
Using 'r' flags makes baby Jesus cry. Stop doing that.
Excuse
Zachary Whitley wrote:
On Sat, 2005-08-06 at 13:02 -0400, Doug Lytle wrote:
Andrew Kohlsmith wrote:
On Friday 05 August 2005 21:31, Doug Lytle wrote:
exten = s,1,Dial(SIP/PHONE1,15,rt)
exten = s,2,Dial(SIP/PHONE4,15,rt)
Using 'r' flags makes baby Jesus cry. Stop doing that.
Thank you all for the extensive help for getting me started on the dialplan.
- Arik
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To UNSUBSCRIBE or update options visit:
No real start, Channel ends and the following is assumed to be the next channel.
On 8/6/05, Zachary Whitley [EMAIL PROTECTED] wrote:
On Sat, 2005-08-06 at 11:36 -0500, Larry Shields wrote:
Check your zapata.conf file. Your terminal profile options section under
[channels] should be ended by
Zachary Whitley wrote:
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost
Using 'r' flags makes baby Jesus cry. Stop doing that.
Excuse me?
r: Generate a ringing tone for the calling party, passing no audio
from
the called channel(s) until one answers. Use with care and don't
insert
this by default into all your dial statements as you are killing call
Robert Goodyear wrote:
Using 'r' flags makes baby Jesus cry. Stop doing that.
Excuse me?
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as
Eric Wieling aka ManxPower wrote:
Robert Goodyear wrote:
Using 'r' flags makes baby Jesus cry. Stop doing that.
Excuse me?
r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by
On Sat, 6 Aug 2005, Robert Goodyear wrote:
Can you educate us all on the appropriate circumstances in which to
use 'r'?
Some devices (voip phones, softphones) do not generate in band progress
information when ringing. You will quickly find out if a particular
end device requires the 'r'
Hello,
I'd like to use g729 pass-thru when I dial out to a sip provider from my
IP phone but because I have no license for g729 I'd like to use g711 ulaw
for asterisk voicemail, conference bridge and other services.
When I set in [general] section of sip.conf the following:
disalow=all
Peter Svensson wrote:
On Sat, 6 Aug 2005, Robert Goodyear wrote:
Can you educate us all on the appropriate circumstances in which to
use 'r'?
Some devices (voip phones, softphones) do not generate in band progress
information when ringing. You will quickly find out if a particular
end
On Sat, 2005-08-06 at 12:34 -0500, Andrew Latham wrote:
No real start, Channel ends and the following is assumed to be the next
channel.
Ok, so the scope of the configuration is from channel= to channel=
statement with the configuration for the channel coming before the
channel statement.
yuppers...
On 8/6/05, Zachary Whitley [EMAIL PROTECTED] wrote:
On Sat, 2005-08-06 at 12:34 -0500, Andrew Latham wrote:
No real start, Channel ends and the following is assumed to be the next
channel.
Ok, so the scope of the configuration is from channel= to channel=
statement with
Ok, so the scope of the configuration is from channel= to channel=
statement with the configuration for the channel coming before the
channel statement.
As in...
these=are
configs=for
the=first
channel=1
these=are
configs=for
the=second
channel=2
In fact, all the settings that
I'm currently running asterisk to provide VoIP services to clients of
the ISP I work for.
I would like to be able to tell if I am loosing packets and/or are
having other issues with any of the voice streams, so I can address them
proactively.
I'm not particularly interested in spending
Hi Ariel.
I having working a Asterisk server with a digital Telmex Line. (10 Lines )
this are my files configurations i hope this can help you
zaptel.conf
span=1,1,0,cas,hdb3
cas=1-15:1101
dchan=16
cas=17-31:1101
loadzone = us
defaultzone=us
unicall.conf
loglevel=255
protocolclass=mfcr2
Frank Tarczynski wrote:
What is the easiest way to check if the H.323 code is working? I've
edited the h323.conf and extensions.conf files but I'm sure that things
aren't right. I've tried connecting to my asterisk box via netmeeting
but I'm having much success. I don't know if my conf
In article [EMAIL PROTECTED],
Jim Duda [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog
phones. The analog phones with the Sipura seem to work great. Voice
quality is fine on both ends on the Sipura. I'm using the
I have my first Zultys here but can't seem to get to first base. Does
anyone have a basic configuration file they might share? Mine is a Zip
2+.
If I understand the Zultys User Manual, the default passwd should be a
null string--but the web interface isn't accepting that (nor 'admin' nor
a couple
Zachary Whitley wrote:
On Sat, 2005-08-06 at 16:14 +0600, Madhawa Jayanath wrote:
Kumara Jayaweera wrote:
Hi all,
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success
stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4).
Please any comments?
I have been trying to setup faxing with a recent CVS-HEAD. I have downloaded and compiled spandsp-0.0.2pre18 and gotten apps_makefile.patch, app_txfax.c and app_rxfax.c
I'm not suprised that the patch failed. Does anyone know what changes need to be made for this to work?
I have very little
If the customers are using an ATA or other VOIP device that
supports RTCP, then you can often get packet loss and jitter stats
by
extracting the RTCP packets and analyzing them.
This will actually give you the packet loss and jitter that
the customer is seeing in the received RTP stream from
Newbie to Asterisk
I've been looking around for a little while, can't seem to find some sample
configs for using a Cisco 7206 as a gateway. The below link is an initial
plan of an Asterisk solution that may replace our Cisco Call Manager 3.1/
IPCC / IVR setup. We currently have all of the
On Sun, 2005-08-07 at 02:54 +0600, Madhawa Jayanath wrote:
Zachary Whitley wrote:
On Sat, 2005-08-06 at 16:14 +0600, Madhawa Jayanath wrote:
Kumara Jayaweera wrote:
Hi all,
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success
stories? my Intel 865 M'd+
Forrest Christian wrote:
I'm currently running asterisk to provide VoIP services to clients of
the ISP I work for.
I would like to be able to tell if I am loosing packets and/or are
having other issues with any of the voice streams, so I can address
them proactively.
I'm not particularly
I am spinning with this for a while. This is a new server that I built
with AAH 1.3 with oh323 support. When I call other parties with my IP
phone, I can talk just fine however I am not hearing anything from
astersik. I ran the server in debug and noticed this :
Aug 6 17:04:43 DEBUG[1429]: Set
On Saturday 06 Aug 2005 22:41, Erick Johnson wrote:
I have been trying to setup faxing with a recent CVS-HEAD. I have
downloaded and compiled spandsp-0.0.2pre18 and gotten apps_makefile.patch,
app_txfax.c and app_rxfax.c
I'm not suprised that the patch failed. Does anyone know what changes
On Sat, 2005-08-06 at 20:55 -0400, [EMAIL PROTECTED] wrote:
As I recall, should channels start as channel=2 and not channel=2?
I have all mine config'ed channel = 2 and it works fine...
Greg
That's correct. My mistake.
___
Asterisk-Users
As I stated before, this is my first system. I'm trying to provide useful
input but I'm not sure what to include. I didn't think you all would want
to see config after config after config. I'll gladly supply config files,
I'm just not sure which ones are relevent.
The first one requested is
I'm starting to see other postings now in other forums (on Digium and AAH)
about this same issue. It's starting to look possibly like an AMP problem,
but there is nothing in their forums about it.
If I find anything, I'll be sure to post here.
Mike
- Original Message -
From: Mike
Hi MARK,
Thanks a lot for the reply. my box is Intel based. and there is no USB
conflicts at all. I ran FC3 well, but, I think new kernel (in the FC4) may
be the place to see.
Thank you
Kumara
- Original Message -
From: MF Hulber [EMAIL PROTECTED]
To: Kumara Jayaweera [EMAIL PROTECTED];
]: -- Executing NoOp(Zap/1-1, SIPUSERAGENT=)
in new stack
Aug 6 23:42:33 VERBOSE[1988]: -- Executing NoOp(Zap/1-1,
TIMESTAMP=20050806-234233) in new stack
Aug 6 23:42:33 VERBOSE[1988]: -- Executing NoOp(Zap/1-1, TXTCIDNAME=) in
new stack
Aug 6 23:42:33 VERBOSE[1988]: -- Executing NoOp(Zap/1-1,
UNIQUEID
On Sunday 07 August 2005 04:05, Brian West wrote:
What are the advantages of using woomera IAX2 instead of native IAX2?
Put woomera aside right now, This is something that brings a cross
platform IAX2 stack that can for example be used in Gnomemeeting or
anything else that uses OPAL, using a
im using iptel.org SER proxy.
the proxy is working without authentication.
the problem is that the Asterisk is not sending a
REGISTER sip message.
--- Juan Salas [EMAIL PROTECTED] escribió:
Which SIP proxy are you using?
Check the authentication parameters (user-id,
auth-id, password)?
Hello.
I am having issues with my amp 1.3 installation after I installed
oh323 support. I do not hear any voice from asterisk. it seems like
whatever codec I choose on my IP phone this is the issue. I can call
other endpoints without problem however.
The debug log shows,
Aug 7 00:45:08
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