On Sat, Aug 13, 2005 at 10:17:17PM -0700, Paul Mahler wrote:
Note that the single line card will not work in a variety of more recent
servers including Dell servers.
First, you have to get the card configured to run with Linux. This means
loading the correct driver and then configuring the
On Sat, 13 Aug 2005, Jamin W. Collins wrote:
Is there a way to initiate a transfer using an analog handset? For
instance I'm looking for a way to do something like the following:
External call comes in and is answered by user A. After talking to the
caller they determine that the caller
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
The problem is not sound setup related. It present even if microphone is
disconnected.
To repeat the question from Matt Riddell:
Does he have Stereo Mix selected as a recording source?
We have found the most common cause of a strong echo to
Hi,
I am using SIP phone (Polycom 300). Echo is present even if other party has
sound hardware disconnected.
It is definetely network and/or PC setup issue, but is not related to audio
setup. I will check stereo mix, however.
Rudolf
- Original Message -
From: Peter Svensson
You don't get 'echo' on the network, you'd only get true echo
connecting to analogue PSTN lines so as Matt pointed out it will sound
set-up/card related. What you could be getting is feedback or sidetone
- so check for things like mic boost and turn that off and it may even
be worth trying another
Damon Estep wrote:
When executing: Dial (SIP/[EMAIL PROTECTED],60
mailto:SIP/[EMAIL PROTECTED],60) I get about 15 seconds of
ringing, the called party rings, but if not answered in the ~15 seconds
I get back SIP 480 temporarily unavailable.
If the call is answered everything is fine
Michiel van Baak wrote:
On 15:14, Sat 13 Aug 05, John Fawcett wrote:
I have successfully configured asterisk to make outgoing calls over FWD,
but cannot receive incoming calls. The console shows no messages,
even though an XTEN client on the same network has no problems receiving
incoming
Hello
I am (attempting) to run the astlinux version of Asterisk on a VIA embedded
platform. I have a TDM04B and pretty sure zaptel.conf and zapata.conf setup
OK. They worked fine with same card in traditional PC anyway.
I think need the module wcfxs for a Digium TDM04B card. Is this
On Sun, Aug 14, 2005 at 11:37:00AM +0100, Angus Comber wrote:
Hello
I am (attempting) to run the astlinux version
Which version?
of Asterisk on a VIA embedded
platform. I have a TDM04B and pretty sure zaptel.conf and zapata.conf
setup OK. They worked fine with same card in traditional
I am running astlinux 0.2.8 - ie latest latest version.
OK so wctdm is alias to same as wcfxs. But even if I load that, it loads OK
pbx sbin # lsmod
Module Size Used by
binfmt_misc 12296 1 - Live 0xde839000
wctdm 129216 0 - Live 0xde855000
zaptel 235844 1 wctdm, Live
P Carballo wrote What does the CLI say? Does it show Playing 'value-of-MES' (language 'en')? I'm using 1.0.8 here and I have no problems using A(x) in my dial strings in either ZAP or SIP channels. Yes, it does say Playing ... (language 'en') but there is no sound sent tothe called
Thanks for reply.
You don't get 'echo' on the network, you'd only get true echo
connecting to analogue PSTN lines so as Matt pointed out it will sound
set-up/card related.
Yes, this would be the logical conclusion, although it is hard to beleive
given what I hear.
It sound like I am talking
Hi, Does anyone know the optimum sound quality to use for playing recorded
voice prompts with Asterisk? What is the best format, quality, etc?
Thanks,
Sam
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Geoff wrote:
Hi, Does anyone know the optimum sound quality to use for playing recorded
voice prompts with Asterisk? What is the best format, quality, etc?
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files
Doug
___
Thanks - always intetested in cures to the dreaded four letter word 'echo' !!
Regards
Rob
On 8/14/05, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote:
Thanks for reply.
You don't get 'echo' on the network, you'd only get true echo
connecting to analogue PSTN lines so as Matt pointed out it
Thanks Doug. Perfect.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Sunday, 14 August 2005 10:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sound Quality?
Geoff wrote:
Hi, Does
Hello there.
What I'm trying to make is - have an asterisk server with
sccp/mgcp/skip/h.323 support to handle calls
between various company locations. Let's say the company has
five different locations, internet connections
in each one of them and would like to use it via asterisk to
IPManager is now fully customizable as everything is generated from
templates using the IPManager database. Complete set of templates is
included in the download. You can configure the dial plan and other
configuration files exactly like you want, using nothing but notepad.
There's a very easy
Just a note to those of us with the SPA-3000 that have had a hard
time with the 'volume' of calls from the PSTN (we're on the line
they're on the PSTN). Anyway I change the parameter:
PSTN To SPA Gain
to 6 and the volume is fine (it's under the PSTN Line tab). I don't
remember seeing this on
Hi; I've been using Asterisk for a few months now, and I have run into
an interesting issue that I thought someone else in the community may
have run into:
I have an Asterisk install set up to receive helpdesk calls, route
them to several IAX extensions and an extension which is simply a
My DID with Telasip is disconnected and my Asterisk box wont
register with them. Anyone else having problems with them?
Jeff
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On Friday 12 August 2005 00:06, Dustin Knuttgen wrote:
Greetings,
I am having a funny problem where a user is on the phone with someone
that dialed in. Another call will come in, the first is put on hold, the
second answered. When the call is transferred the two external calls are
connected
Dear forum,
I have a install of asterisk using AMP. I followed the install guide
off the AMP site.
http://amp.coalescentsystems.ca/docs/AMP_Installation_Guide_v1.4.pdf
When I start using amportal start or asterisk -v I received this in my log.
The last line is that ogg failed. I have
I'm trying to implement a shared asterisk server for multiple
(different) companies. Here's what I've done so far:
- I've installed multiple asterisk instances on one server (via
vserver). Each * is for one customer, and has it's own extensions (like
100, 101, 102, etc.) Note that the same
ogg_vorbis is a media format, like MP3/WAV/etc (most notably used for
music on hold in the case of Asterisk).
You are probably getting this error because you're missing required
libraries on your linux install in order for the module in asterisk to
be able to read ogg_vorbis files.
If you
Damon Estep wrote:
When executing: Dial (SIP/[EMAIL PROTECTED],60
mailto:SIP/[EMAIL PROTECTED],60) I get about 15 seconds
of
ringing, the called party rings, but if not answered in the ~15
seconds
I get back SIP 480 temporarily unavailable.
If the call is answered everything is
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote:
You don't get 'echo' on the network, you'd only get true echo
connecting to analogue PSTN lines so as Matt pointed out it will sound
set-up/card related.
Yes, this would be the logical conclusion, although it is hard to beleive
given what
Thanks Luki,
Seems easy enough, does the code look like it would be hard to change
that value from a hard coded value to a global variable which can be
defined in voicemail.conf and overridden for a single mailbox?
I am not a coder so an opinion would be useful.
I have cross posted this to
I have a different problem, but you might end up using the same
solution as I did. I wanted to change to the cmu_us_slt_arctic_hks
voice (which sounds AMAZING to me.. way better than what ATT wants 300
bucks for), but no matter what I did, I couldn't get Asterisk to work
with it. So I am using a
- I've installed multiple asterisk instances on one server (via
vserver). Each * is for one customer, and has it's own extensions (like
100, 101, 102, etc.) Note that the same extension can exist on other *
instances
This is completely UNNECESSARY if you simply use contexts. We have 1
Jeff Borders wrote:
I think I have a bad FXS module on my TDM400P.
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
I remember
subscibe FOODFIX digest
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I have ParkAndAnnounce set up to call my cell instead of Console/DSP and
this works fine. If a caller tries to get me on my cell, they are
parked... I am called and told where they are holding, but then
ParkAndAnnounce hangs up. So I have to call my system back and dial the
parked extension to
On 8/13/05, Tom Rymes [EMAIL PROTECTED] wrote:
snip sean
This is not that difficult. The first question you need to answer is
whether you want to use a dedicated circuit (E1/T1/PRI) or multiple
copper lines. This mostly depends on the call volume that you will be
handling. Depending on
Hi,
On 11:41, Sun 14 Aug 05, Matthew Boehm wrote:
- I've installed multiple asterisk instances on one server (via
vserver). Each * is for one customer, and has it's own extensions (like
100, 101, 102, etc.) Note that the same extension can exist on other *
instances
This is
Hey all!
Have configured a Cisco 7960 with no problems, put up an
TFTP server and it downloaded new sip binaries all went well.
However, now I am having trouble getting two 7040s to work. Basically,
my problem is the above stated error message. If I had any entries in the TFTP
Regarding my earlier email (for some reason, I don't get my own emails
from the list), I looked at the code and although I'm no programmer, I
see that this is meant to hangup after the announcement. If I comment
that line out, the call remains on the line, but I'm in limbo. I tried
to add a
On Sunday 14 August 2005 14:17, Michiel van Baak wrote:
This is completely UNNECESSARY if you simply use contexts. We have 1
asterisk server running 6 different companies and a good majority of
their extensions overlap. This is very easy to configure.
Do you also have some kind of tool
On 20:23, Sun 14 Aug 05, Bjorn Ove Kristiansen wrote:
Hey all!
Have configured a Cisco 7960 with no problems, put up an TFTP server and it
downloaded new sip binaries ? all went well.
However, now I am having trouble getting two 7040s to work. Basically, my
problem is the above stated
Thank you for your reply BJ,
I have check my yum list and I have all kinds of libs for oggvorbis.
I will give it a try.
Thank you for your time,
Tommy Denton
[EMAIL PROTECTED]
On 14/08/05, BJ Weschke [EMAIL PROTECTED] wrote:
ogg_vorbis is a media format, like MP3/WAV/etc (most notably used
Almost.. A call on hold doesn't represent the true bandwidth and CPU that a
*real* call utilitizes. Short of producing an echo or feedback on each call
to make it look like a real call, I'm not sure how you could create a real
call test scenario.
-Original Message-
From: [EMAIL
If they are all SIP based devices why not use the standard G711. The
server will only be doing real work when one of the users want to make
an outside call.
As a guide, I run an Dual HP DL380 2.8GHz with 1GB of RAM and a Quad T1
card with 3 PRI's attached for a total of 120 users 30 of which
Ok,
After following BJ's advice and removing ogg.so I then got a
pbx_realtime.so error in the same fashion. I removed that file, and
then the next and then the next as you can see in the log below.
I think something is not right. duh here is my sign..lol...but I am
not sure even where this
Using the install instructions for [EMAIL PROTECTED], I setup a FWD account,
this I
tested using X-Lite and it works okay,
Nowever I cannot make calls to fwd using Asterisk, my log showes:
Aug 14 21:06:09 NOTICE[1324]: Registration of '689482' rejected:
Registration Refused
Aug 14 21:06:59
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, August 14, 2005 12:00 PM
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users Digest, Vol 13, Issue 98
Send Asterisk-Users mailing list submissions to
Do you also have some kind of tool so the companies can
manage their own context ?
Yup. We use RealTime Extensions. Customers login to a password protected
website. User/pass is tied to their context so they are able to
add/delete/update anything in their extensions context.
We tell them
On 8/13/05, Tom Rymes [EMAIL PROTECTED] wrote:
snip sean
This is not that difficult. The first question you need to
answer is
whether you want to use a dedicated circuit (E1/T1/PRI) or multiple
copper lines. This mostly depends on the call volume that
you will be
handling.
Matthew Boehm wrote:
Do you also have some kind of tool so the companies can
manage their own context ?
Yup. We use RealTime Extensions. Customers login to a password protected
website. User/pass is tied to their context so they are able to
add/delete/update anything in their extensions
Michiel van Baak wrote:
I have put this in my dhcpd.conf to make sure my cisco
phones connect to my TFTP server:
server-name 192.168.2.1;
I'd be surprised if that worked... the server name is for.. um.. the
name of the server :)
Try:
option tftp-boot-server code 150 = ip-address;
option
Hello Jason,
I've just come across your post to the Asterisk-Users group
regarding OrderlyQ (from a web search - sorry it's taken so long).
You wrote:
What experience can be shared about installing and running the
OrderlyQ application? I have a bunch of
queues set up and want to start
hi
anyone here that knows a good howto of setting up rate-engine?
i've made some changes to make it work on cvs head, but the
documentation is rather poor, so i just wonder..
roy
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Thats a good idea...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Darren Wiebe
|Sent: Sábado, 13 de Agosto de 2005 12:43 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Load Testing
|
|I just
You're right. Sounds like there's definitely something more going on
that just a missing library. Where did you get this version of
asterisk from? The CVS server or a binary/RPM installed from
somewhere? Is this the first install on this machine or has there been
other installs prior to this?
On
BJ Weschke wrote:
You're right. Sounds like there's definitely something more going on
that just a missing library. Where did you get this version of
asterisk from? The CVS server or a binary/RPM installed from
somewhere? Is this the first install on this machine or has there been
other
CVS both times.
First I ran from the directions on the offical documentation project,
then I wanted to us amp as I am not going to be arround much longer
and the folks that need to do the dial plans need gui..
should I scrap and go again?
Thank you for your time,
Tommy
On 14/08/05, BJ
Tommy Denton wrote:
CVS both times.
Once from CVS-STABLE and once form CVS-HEAD I'd say.
First I ran from the directions on the offical documentation project,
then I wanted to us amp as I am not going to be arround much longer
and the folks that need to do the dial plans need gui..
AMP
Quick and dirty way would be to then dump you into DISA and then retrieve
the call from the parking lot
Just a thought
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris Edwards
Sent: Sunday, 14 August 2005 10:37 PM
To: Asterisk Users
Could someone tell me if the h priority is limited to one per dial plan
or is it one per context?
Thanks,
Doug
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Do i have to create voice mail one by one (per ext at
the meridian) at the * box ?
--- Mark Phillips [EMAIL PROTECTED] wrote:
Easily doable. I've done it twice now. Problem is
that your users will
never know they have messages waiting.
Install a T1/E1 card into the * box and then use a
Thanks. It works correctly with this change.
Kun
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matthew
Boehm
Sent: Friday, August 12, 2005 9:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] app_voicemail.c
anyone use TE4xxP work well with huawei CC08 switch?
DO YOU YAHOO!?
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David Phelan wrote:
Quick and dirty way would be to then dump you into DISA and then retrieve
the call from the parking lot
Just a thought
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris Edwards
Sent: Sunday, 14 August 2005 10:37 PM
Tim Connolly wrote:
On that note... IPSec tunnels seem to reek havoc on the echo
canceling/training process. Anytime our Cisco PIX loads up, the echo
complaints start coming in. Stay away from the IPSec tunnels.
Any idea why that would happen? Obviously on a 4-wire end-to-end system
echo
so is there anyway to just delete the lot? or do I have to drive to
the colo and reinstall..
I know this answer, the sys admin in my body tells me that when you
have messed up this bad you are better to just gack and load it up
again...but alas I will take my beating..
Thank you for your help,
rm -rf /usr/lib/asterisk/modules/
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Title: RE: [Asterisk-Users] yet another Asterisk and VMware question
I did find this on the VMware wish list re Asterisk and PCI slots. I'm not holding my breath.
http://www.vmware.com/community/thread.jspa?threadID=7839=65462
I guess I'll drum up an PC from spares.
-Original
On Sat, 2005-08-13 at 19:53 -0400, Jeff Buchbinder wrote:
Hi; I've been using Asterisk for a few months now, and I have run into
an interesting issue that I thought someone else in the community may
have run into:
I have an Asterisk install set up to receive helpdesk calls, route
them to
That is it..then just start over? I don't need to kill the source as well?
Do I need to get the /etc/asterisk directory as well?
Tommy
On 14/08/05, William Suffill [EMAIL PROTECTED] wrote:
rm -rf /usr/lib/asterisk/modules/
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On Fri, 2005-08-12 at 08:29 -0500, Kevin P. Fleming wrote:
Matt Florell wrote:
We do 2nd gen firmware upgrades for customers every day, have been doing
them for a couple of weeks now. The process is simple, just contact the
RMA department and be prepared to pay for shipping the card to and
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app_meetme2.c:646: error:
'__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared
here (not in a function)make[1]: *** [app_meetme2.o] Error 1make[1]:
Leaving directory `/usr/src/asterisk/apps'make: *** [subdirs] Error
1
I need a
copy of the app_meet2.c file that
You can also remove /etc/asterisk to erase the configs that were
installed but the major issue between STABLE/HEAD is the modules. The
version mismatch in the modules is what caused all the errors you got
such as Aug 14 15:04:33 WARNING[4860]:
/usr/lib/asterisk/modules/app_realtime.so: undefined
It worked!! I have removed and reinstalled..things are working!! I
might just make a phone call tonight!!
Thanks ALL!!!
On 15/08/05, William Suffill [EMAIL PROTECTED] wrote:
You can also remove /etc/asterisk to erase the configs that were
installed but the major issue between STABLE/HEAD is
Okay,
First of all, thank you for your input. I didn't know that I could use 1
* for multiple companies (wish I knew it earlier, because installing
vserver and installing * on a vserver took me a lot of time :) ).
Nevertheless, I think I still will need the SER. If my 'shared *' server
is getting
Hi All,
Can Asterisk dial extension which resides in the PABX?
(eg. 2000) Sip Phone - Asterisk -- ATA (FXS) -- (CO
side) PABX - Extension (eg. 1000)
(2100 2101)
can my sip phone call to pabx extension
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