On Mon, 2006-10-23 at 00:48 -0500, Eric ManxPower Wieling wrote:
Voicemail allows the caller to leave voicemail. Voicemailmain allows
you to check your voicemail.
I got this one.
1.0.x Asterisk mailbox options were put as a prefix to the mailbox,
such
as Voicemail(u11) would play the
Hi Has anyone used the AudioCodes MP-20x?I've been testing this for 3 weeks now. No problems so far. This gateway has many features including IPSec and is not that expensive.
RegardsAndrew
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Great!
Thanks for your aid... I spend a lot of day around this problem...
Now realtime load returns data!
- Original Message -
From: Tijl Van den Broeck [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October
Alls well that ends well !!! :-)
Maurizio Pederneschi wrote:
Great!
Thanks for your aid... I spend a lot of day around this problem...
Now realtime load returns data!
- Original Message -
From: Tijl Van den Broeck [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
So, What´s your recommendation for a production environment? I waslooking for good prices, good voice quality for USA Origination and I´d
like to hear about good experiences
PSTN. Just can't beat the quality :-) Wait, you said good prices. Sorry.
On 10/23/06, Andrew Nowrot [EMAIL PROTECTED] wrote:
I've been testing this for 3 weeks now. No problems so far. This gateway has
many features including IPSec and is not that expensive.
Appreciate if you can post the sample configs to wiki or to the list.
There is no information about
Is it possible to compile the h323 channel before installing
the full asterisk package as mentioned within the README
Within the Asterisk/channels/h323 Directory
It compiles after asterisk has been installed, but no
chan_h323.so has been created within the channels directory
Been
Any more ideas, esp from guys whove used this in their setp?
Benjamin Jacob wrote:
Giovanni,
Appreciate your lines mate.
But, Ive already read those, all over the net.
my qs inline :
amaflags : Categorization for CDR records. Choices are default, omit,
billing, documentation and choices
After a reboot, asterisk is usually too much in a hurry to try and resolve
my iax/sip providers.
Asterisk starts before the internet connection is up and dns is working.
Then asterisk just waits, and waits and waits and waits even longer before
ever trying to revolve any voip provider again.
Hello All!
At start of an asterisk I see the following:
== Primary D-Channel on span 2 up
== Primary D-Channel on span 2 down
Oct 23 10:26:25 WARNING[9515]: chan_zap.c:2287 pri_find_dchan: No
D-channels available! Using Primary channel 47 as D-channel anyway!
== Primary D-Channel on span
Remco,
Asterisk starts before the internet connection is up and dns
is working.
knip
And then people say nightly asterisk restarts are not a good idea
Why is your asterisk startup script running before networking has been
setup? Asterisk has the same networking dependencies as
Hi,
Any suggestions to below problem?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jamie
Heckford
Sent: 17 October 2006 21:48
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FW: Why does One-Touch record mute/disconnect
callif
Hi, I'm using Asterisk with a Digium TDM10B FXO card and it's driving me
nuts. I'm based in the UK and have echo problems and need to switch the
driver from FCC mode to UK mode.
I've tried modprobe zaptel and modprobe wctdm opermode=UK and the ztcfg. I
get no error messages but when I reboot it
On 10/20/06, Mohammad Shokuie [EMAIL PROTECTED] wrote:
Anyways, let me take the most benefit as im sure you'd read this post, i
have problem with the size of received page which is shrinked, can u give me
a hint about this problem too :)
This is probably the problem of the application that
On Mon, 2006-10-23 at 10:14 +0200, Andreas Sikkema wrote:
Why is your asterisk startup script running before networking has been
setup? Asterisk has the same networking dependencies as apache, so I
start it around the same time using the same priority as apache and as
far as I know
Hi,
Could anyone knows whatwent wrong with theerror below result of installation of libsupertone.
[EMAIL PROTECTED] latest]# tar xvf
Looking at the source code for Asterisk 1.2.7.1 (just what I've got
handy), it appears that the SIP_HEADER() function just parses the SIP
INVITE for whatever SIP *header* you specify - so:
a) there's no list of headers you can check for - it depends on the user
agent generating the request and
b)
On Monday 23 October 2006 21:45, Angel Heart wrote:
Hi,
Could anyone knows what went wrong with the error below result of
installation of libsupertone. [EMAIL PROTECTED] latest]# tar xvf
libsupertone-0.0.2.tar
[snip]
libsupertone-0.0.2/aclocal.m4
[EMAIL PROTECTED] latest]# ./configure
Hi,
Thank you for your comment;
Below was the result of ./configure
checking how to run the C++ preprocessor... /lib/cppconfigure: error: C++ preprocessor "/lib/cpp" fails sanity checkSee `config.log' for more details.[EMAIL PROTECTED] libsupertone-0.0.2]# Please comment.
Thanks again.
-
On Mon, Oct 23, 2006 at 09:34:10AM +0100, Neil Tancock wrote:
Hi, I'm using Asterisk with a Digium TDM10B FXO card and it's driving me
nuts. I'm based in the UK and have echo problems and need to switch the
driver from FCC mode to UK mode.
I've tried modprobe zaptel and modprobe wctdm
On Mon, Oct 23, 2006 at 02:11:22AM -0700, Angel Heart wrote:
Hi,
Thank you for your comment;
Below was the result of ./configure
checking how to run the C++ preprocessor... /lib/cpp
configure: error: C++ preprocessor /lib/cpp fails sanity check
See `config.log' for more details.
[EMAIL
Hello friends,
I am getting this error:-
Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value
'192.168.1.12:5060:300:15553695861:sip:[EMAIL PROTECTED]:5060' for key '23' in
family 'SIP/Registry
I have no idea what it means. Please tell me what could be the problem.
With
Minor update - use the following:
if (strcasecmp(data,
x-Asterisk-Request-URI-pseudo-header)==0)
{
ast_copy_string(buf, p-initreq.rlPart2, len);
-Original Message-
From: Steve Langstaff
Sent: 23 October 2006 09:58
To:
On Mon, 23 Oct 2006, Andreas Sikkema wrote:
Remco,
Asterisk starts before the internet connection is up and dns
is working.
knip
And then people say nightly asterisk restarts are not a good idea
Why is your asterisk startup script running before networking has been
Hi,I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get error: configure: error: Can't build without libtiff . But I have installed tiff from port tiff-3.8.2. I understand that the problem is about libtiff, and spandsp can't find these libs. So how to fix the problem?
Thanks
I checked the file permissions. They are proper. There doesnot seem to be a
visible error. No change has been done in any conf files for the past 4 months.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
Soft phones or hard phones ? For softphones you can just use the PC's. If
you go with hard phones you may want to get phones with QOS or build a
seperate network for the phones.
- Original Message -
From: Bo Yang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday,
Title: Zap Channel and VM problem
Hello,
I am experiencing problems with a ZAP channel not being released after a voicemail has been left.
I have 2 analogue extns from an Alcatel PBX wired into 2 FXO ports on my * box.
If I make a call from an Alcatel extn and route it to a SIP
On 10/23/06, Giedrius Augys [EMAIL PROTECTED] wrote:
Hi,
I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get error:
configure: error: Can't build without libtiff . But I have installed tiff
from port tiff-3.8.2. I understand that the problem is about libtiff, and
spandsp can't
On Mon, Oct 23, 2006 at 12:52:49PM +0100, Andy Green wrote:
Hello,
I am experiencing problems with a ZAP channel not being released after a
voicemail has been left.
I have 2 analogue extns from an Alcatel PBX wired into 2 FXO ports on my *
box.
If I make a call from an Alcatel extn and
Hi All
I would greatly appreciate some advice or some direction as
to where to go next.
I have a provider passing me incoming calls via my Session
Border Controller.
I am able to pass them calls fine but coming in fails with a
407 Authentication Fail error.
In my sip.conf I have
I'm investigating in deploying an asterisk solution. After some experiments, I noticed that asterisk is quite subect to unreproducible troubles and quality losses with variations
of server load, and also a lot of troubles that could be reconducted to timing problems, expecially with faxes.
I was
Hi!
This weekend we had a problem with our Asterisk Box which ran flawlessly
for nearly 4 weeks. The Asterisk server sits between the PSTN and a
Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not
outgoing. The admin rebooted the Dell Box and then everything worked
fine again.
Joseph wrote:
Though what option am I suppose to pass it.
The process seems to me correct, when I get-in to disa-access I have
access to voicemail extension 1000 (otherwise it wouldn't let me dial
ext. 1000; when I dial it it asking me for mailbox number and password,
except that password is not
Hi Tzafrir,
If I do a cat of /sys/module/wctdm/opermode I get:
FCC
I thought I could change it here and then do a ztcfg or a genzaptelconf but
it just overwrites it with FCC again.
How do I change it?
Neil
safeharbour IT Ltd
Your IT Department
tel: 0845 644 3607
fax: 0845 867 2891
mob:
Ok didn't know TrixBox had remote, control support ...
If anyone has please tell ..
Thanks
greg
On 10/20/06, Matthew Rubenstein [EMAIL PROTECTED] wrote:
Has anyone used the TrixBox/AAH builtin facility xPL for
facility (including home/office/industrial) automation?
On Fri, 2006-10-20
2006/10/23, Steve Davies [EMAIL PROTECTED]:
On 10/23/06, Giedrius Augys [EMAIL PROTECTED] wrote: Hi, I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1.I get error: configure: error:Can't build without libtiff . But I have installed tiff
from port tiff-3.8.2. I understand that the
You might want to repost it with a subject or you miss a lot of people seeing or opening it up.
-- Original message -- From: "Scott Pinhorne" [EMAIL PROTECTED]
Hi All
I would greatly appreciate some advice or some direction as to where to go next.
I have a provider
Hi Klaus.
I'm not sure about the timer expiry meaning,
but you could use the xlog command (usually found
in /usr/lib/eicon/divas)
Just run it as root indicating which span (1..4)
you want to trace:
./xlog -c 2
that shoud show you layer 1 layer 2
dump
Alberto.
Klaus Darilion ha scritto:
Good prices means (exactly) reasonable prices. I´m a newbie, so I´m
asking for good experiences...
Thanks in advance...
R.R. Libera
Lacy Moore - Aspendora escribió:
So, What´s your recommendation for a production environment? I was
looking for good prices, good voice quality for USA
Remco Barendse wrote:
It is not, asterisk is correctly started after networking services,
however it seems that when the box is booting the dns is replying just a
split second too late for the taste of asterisk and it seems that asterisk
then marks the provider as unavailable.
* should
Hello,
I've posted this at the trixbox and freepbx forums and haven't been able
to get an answer. I thought perhaps the guru's here might be able to
help me out :)
I'm having some issues with setting caller IDs. There are 2 problems
that I would like to solve.
1. I have a DID pointing to
On Mon, Oct 23, 2006 at 02:32:55PM +0300, Giedrius Augys wrote:
Hi,
I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get
error: configure: error: Can't build without libtiff . But I have
installed tiff from port tiff-3.8.2. I understand that the problem is
Grab the fax2mail script from www.generationd.com and set it to convert the
tiff to pdf before sending. Works great.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: Monday, October 23, 2006 4:38 AM
To: Asterisk Users Mailing List
Greetings, All!
Help to find the reason, constantly writes: == Primary D-Channel on span
2 up
---
Log:
== Primary D-Channel on span 2 up
== Primary D-Channel on span 2 down
Oct 23 10:26:25 WARNING[9515]: chan_zap.c:2287 pri_find_dchan: No
D-channels available! Using Primary channel 47 as
The 7960's have an envelope that appears in the display next to a line
which has voicemail. Also, the MWI light is a logical OR of all the
defined lines.
Is there a way to tell the phone NOT to display the MWI for certain
lines but retain the envelope for all? If you get enough VM on busy
On 10/23/06, mail-lists [EMAIL PROTECTED] wrote:
Hello,I've posted this at the trixbox and freepbx forums and haven't been ableto get an answer. I thought perhaps the guru's here might be able tohelp me out :)I'm having some issues with setting caller IDs. There are 2 problems
that I would like to
I have had some interesting compiling results with the
latest beta release of Asterisk. With reference to this channel
After running the make opt in the H323 directory, and the
make install in the Asterisk directory, there is still no chan_h323.so file
Created.. Are there any other
Hello,
I had the same trouble, it was the telco operator that had an equipment
in fault...
It was unable to get the RNIS level 2 communication up...
You should issue: pri intense debug span 2 and see what happens to the
line...
If you see SABME msg going in and out in loop, it is what i
On Mon, Oct 23, 2006 at 01:46:53PM +0100, Neil Tancock wrote:
Hi Tzafrir,
If I do a cat of /sys/module/wctdm/opermode I get:
FCC
I thought I could change it here
No point in changing it there. It is used when the module is initialized
(and should be a read-only parameter, hint-hint)
- Original Message -
From: mail-lists [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 23, 2006 3:39 PM
Subject: [asterisk-users] CID Issues
Hello,
I've posted this at the trixbox and freepbx forums
On Mon, Oct 23, 2006 at 04:09:43PM +0200, Patrick wrote:
I have had some interesting compiling results with the latest beta release
of Asterisk.. With reference to this channel.
After running the make opt in the H323 directory, and the make install in
the Asterisk directory, there is
Ok, I've somehow resolved this. I've added the line options wctdm
opermode=UK to modprobe.conf, rebuilt zaptel and now when it boots I get UK
mode instead of FCC mode and, hey presto, less echo!
Many thanks Tzafrir!
Neil
safeharbour IT Ltd
Your IT Department
tel: 0845 644 3607
fax: 0845
When I have a look and the menuselect.makeopts file.. MENUSELECT_CHANNELS=
chan_gtalk chan_h323 ... is there, and also during the ./configure, all the
various pwlib and openh323 version checks seem valid.. but still not sure
where you enable the channel to be built...
According to the H323
Is there any way under software control (CLI, Manager, etc.) to busy out
one or more PRI channels, for testing purposes, without actually having
to make real calls on them? It should have the following effects:
a) Outgoing calls will not try to use the busied out channels, when using
a group
Can someone tell me if this indicates a problem? What does it mean when a macro
exits != 0 ?
Spawn extension (macro-syst_FindAppServer, s, 5) exited non-zero on
'SIP/xxx.yyy.142.186-b7515f98' in macro 'syst_FindAppServer'
Thanks,
Doug.
___
Thomas Kenyon wrote:
Remco Barendse wrote:
It is not, asterisk is correctly started after networking services,
however it seems that when the box is booting the dns is replying just
a split second too late for the taste of asterisk and it seems that
asterisk then marks the provider as
Thanks..
Did I misread the posts? These look like the VWIC-1MFT-T1 is connecting to
the PSTN and not connecting to a PRI card on an Asterisk box..
We are looking to do the following..
Asterisk PRI card - VWIC-1MFT-T1 - SIP -
Thanks
David
- Original Message -
From: Tijl Van den
mail-lists wrote:
Hello,
I've posted this at the trixbox and freepbx forums and haven't been able
to get an answer. I thought perhaps the guru's here might be able to
help me out :)
I'm having some issues with setting caller IDs. There are 2 problems
that I would like to solve.
1. I have
David Edwards wrote:
Thanks..
Did I misread the posts? These look like the VWIC-1MFT-T1 is connecting to
the PSTN and not connecting to a PRI card on an Asterisk box..
We are looking to do the following..
Asterisk PRI card - VWIC-1MFT-T1 - SIP -
Why not have Asterisk connect directly to
Hi,
Im working with the following versions:
-asterisk-1.2.12.1
-zaptel-1.2.9.1
And with the following card:
00:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
Subsystem: Unknown device 8085:0003
Flags: bus master, medium devsel, latency 32, IRQ
Hi all,
Does Asterisk now support Intels HMP platforms?
Does it support in 1.4 version ?
Thanks.
Greg
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To UNSUBSCRIBE or update options visit:
Sorry a typo.. Having one of those Mondays..
Non-IP PBX (PRI Interface) - VWIC-1MFT-T1 - SIP - Asterisk
I am considering recommending/testing something like the Quintum Tenor
products.. I like Cisco, but in this case it might not be the best option..
David
- Original Message -
On Mon, 2006-10-23 at 07:46 -0500, Eric ManxPower Wieling wrote:
Joseph wrote:
Though what option am I suppose to pass it.
The process seems to me correct, when I get-in to disa-access I have
access to voicemail extension 1000 (otherwise it wouldn't let me dial
ext. 1000; when I dial it
Joseph wrote:
On Mon, 2006-10-23 at 07:46 -0500, Eric ManxPower Wieling wrote:
Joseph wrote:
Though what option am I suppose to pass it.
The process seems to me correct, when I get-in to disa-access I have
access to voicemail extension 1000 (otherwise it wouldn't let me dial
ext. 1000; when I
Thanks to all that replayed, I made like Mr Watkins told me, and my problem is
apparently solved, although, because of the usage of the syntax
VoiceMail(${EXTEN}|u), now, two more sound files are played: vm-theperson and
vm-isunavail, while before were only played vm-intro and beep.
Is there a
The reason not many people have this product, is because this product is not going to be available to the public at this time.Audiocodes will only provide this product (currently due to ship in December) as 1,000-piece minimum orders for the MP-202. The MP-201 will be available sometime quarter 1
All,
Has anyone seen INVAL messages on an IAX link before?
I'm occasionally gettingthem from my Gateway provider,
and I need to narrow down the potential cause.
Symptoms are: Incoming calls fail, I see NEW,
AUTHREQ then INVAL messages between the two A*k
Thanks for all that replayed, the problem is solved!
Regards,
Ricardo.
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The o option is mentioned over at FreePBX and how to restore this setting.
On 10/23/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
mail-lists wrote: Hello, I've posted this at the trixbox and freepbx forums and haven't been able
to get an answer. I thought perhaps the guru's here might be
[snip]
What I don't understand is why the CALLERIDNUM is pstn1270. Also, you
can see the VoicemailMain is stripping off the ps, I think that may be
because you do not have a | or a , after ${CALLERIDNUM}. Why not
just REMOVE the ${CALLERIDNUM} and let VoicemailMain prompt you for the
Hi list,
I have a call file as following and it works. But, I don't really understand
its mechanism.
The SIP/voipbuster is a sip trunk which I set up in freePBX with voipbuster
account. And 2874 is one of my extension which was assigned to x-lite
client.
When I place this call file in outgoing
Hi Adrian,
are you using this IAX thru NAT? I have this problem when i try call
with IAX2 and this Asterisk server is behind the NAT...
I think its here problem with UDP source port which is changed in NAT
router, but im not sure 100%
Marian
Adrian Marsh napsal(a):
All,
Has
If you dont mind me asking a few questions, I am wondering, to what extent have you tested the units? Do all the basic functions (call id, call waiting, call transfer, forwarding, etc) work on the unit? How well do the router functions work? Overall quality and impressions?
On 10/23/06, Andrew
Second authentication DISA is for additional security Actually that's called paranioa
and it doesn'tcause any problem, Obviously you do have a problem accessing your voicemail, else you would not be posting this
the authentication is giving me access to voicemailbut password is not recognized.
Elvar
Are you looking at a multi-site VoIP system or just replacing a PBX at
a large site. Using VoIP at a multi-site client that does not have
full time service personnel can lead to a failure that affects
business operations. Always verify your COOP before starting such an
install.
Please
Hi,
I have asterisk 1.2.12 running on my server. Everything seems to be
working fine on it. It has an IAX connection to the
terminator/orignator. Again, everything seems to be fine.. calls
come in and go out. However, it seems that after a call has been up
for several minutes audio will go
Anybody with photo's (for this astricon or any asterisk related event),
please upload them at:
http://www.asteriskguru.com/gallery/main.php
It's possible to upload as a guest without registering, if somebody
sees kiddie porn etc, please warn me so that i can disable this.
I will be adding
Hi everybody
Im the IT Manager for a new call center and my bosses has assing to me a very dificult task
i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer
that allows outbound calls.
I triyed using terminal services but it dind worked because the
Hi,
I have asterisk 1.2.12 running on my server. Everything seems to be
working fine on it. It has an IAX connection to the
terminator/orignator. Again, everything seems to be fine.. calls
come in and go out. However, it seems that after a call has been up
for several minutes audio will go
I have been struggling over central provisioning for quite
some time. I have eagerly watched each post with like problems but have yet to
find a reliable answer.
I have a Polycom 501 and I am trying to provision from an
FTP server, and just to take any routing out of the issue it is on
On Mon, 2006-10-23 at 15:34 -0400, Andrew Joakimsen wrote:
Second authentication DISA is for additional security
Actually that's called paranioa
I just try with single authentication DISA, doesn't work, password is
not recognized.
[snip]
the authentication is
It looks like the deal CBeyond is offering me for a T1 to the office and VoIP service via SIP is going to win in my current effort to get away from the local telco. The idea of using a VoIP carrier with QoS all the way between here and there and back is very appealing after working on an ADSL
I'm running Asterisk 1.2.13 on a Solaris 10 X86 box behind an IPCop
firewall on a 5Mbps down/512 up cable connection.
I'm having sound quality problems when users call in for voicemail and
with music on hold. The sound is choppy and muffled while souding pretty
good for calls inside the network.
Do you have the issues locally ? Are you using Ztdummy ?
- Original Message -
From: Frank Tarczynski [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 23, 2006 10:48 PM
Subject: [asterisk-users] Where to best start looking for voicemail/moh
sound quality
Hey all,
Has anyone had any issues with phones having multiple lines that are in
different contexts? We've got a couple phones that we're testing
intercom functionality for, and I'm noticing that for some strange
reason, no matter what line we use, the phones tend to be completely in
one context
Sell the thin clients and their home server on Ebay and buy some used
desktops and an Asterisk server.
VOIP with softphones on thin clients does not work very well at all
unless you seriously limit the number of clients attached to each
server.
How many seats is this supposed to be?
MATT---
100 for start and 400 in february OH GOD !!
On 10/23/06, Matt Florell [EMAIL PROTECTED] wrote:
Sell the thin clients and their home server on Ebay and buy some useddesktops and an Asterisk server.
VOIP with softphones on thin clients does not work very well at allunless you seriously limit
SO THERES NOT WAY TO MAKE VOIP WITH THIN CLIENTS YOU SAID?
On 10/23/06, Ignacio Ortega A. [EMAIL PROTECTED] wrote:
100 for start and 400 in february OH GOD !!
On 10/23/06, Matt Florell [EMAIL PROTECTED]
wrote:
Sell the thin clients and their home server on Ebay and
I am running a call center with 20-30 operators with outbound
projects. We have an Digium Quad port E1 interface (TE410P) on an IBM
Server running Ubuntu-server with the lastest version of Asterisk Zaptel
and Libpri.
The problem is that when there are about 15 or more active calls on
On 10/23/06, Joseph [EMAIL PROTECTED] wrote:
I just try with single authentication DISA, doesn't work, password isnot recognized.Try without any disa whatsoever
[snip] the authentication is giving me access to voicemail but password is not recognized. It's giving you access to the voicemailmain
Do you use a VoIP provider? Is that provider by any chance VoicePulse? Have you tried other providers?You can get free DID's at http://www.ipkall.com/ and
http://www.trxtel.com/ that would be a good place to start if you don't have any more providers...On 10/23/06, Frank Tarczynski
[EMAIL
What if you just use the default configuration files?On 10/23/06, Curt Shaffer [EMAIL PROTECTED] wrote:
I have been struggling over central provisioning for quite
some time. I have eagerly watched each post with like problems but have yet to
find a reliable answer.
I have a
Hi
On Mon, Oct 23, 2006 at 04:08:15PM -0400, Ignacio Ortega A. wrote:
Hi everybody
Im the IT Manager for a new call center and my bosses has assing to me a
very dificult task
i have to configure the call center using Hp 5520 thin clients, asterisk and
some kind of dialer
that allows
Date: Mon, 23 Oct 2006 16:48:09 -0400 (EDT)From: Frank Tarczynski
[EMAIL PROTECTED]Subject: [asterisk-users] Where to best start looking forvoicemail/moh sound quality problem?To:
asterisk-users@lists.digium.comMessage-ID:[EMAIL PROTECTED]Content-Type: text/plain;charset=iso-8859-1
I'm running
You are using bad software to view the faxes. In Windows the picture and fax viewer seems to work fine, however in Linux KGhostView or whever the default program is does not work, however you should try KFaxView.Steve: I'm wondering if one day span_dsp will support T38, say we have a SIP provider
Do you mean .cfg and sip.cfg? Could you clarify for me please and I will try that. Thanks for the suggestion.
Curt
On 10/23/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
What if you just use the default configuration files?
On 10/23/06, Curt Shaffer
[EMAIL PROTECTED] wrote:
I have
i recently bought an SP4000 conference phone but having problem
provisioning it using ftp, every time it just hangs at
Updating initial configuration... screen. when i switch it
to tftp, it'll work fine. i though it was bootrom/firmware issue
so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it
On Mon, 2006-10-23 at 17:59 -0400, Andrew Joakimsen wrote:
On 10/23/06, Joseph [EMAIL PROTECTED] wrote:
I just try with single authentication DISA, doesn't work,
password is
not recognized.
Try without any disa whatsoever
I think DISA has to be there as it gives
Tzafrir Cohen wrote:
*snipped
Note that you better not use a terminal server settings. The SIP client
should run on the thin client's CPU, not on the server's CPU. The server
can help with the boot process (maybe a shared NFS root will prove
useful).
*snipped
that particular unit is also
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