On 4/29/05, Rich Adamson [EMAIL PROTECTED] wrote:
I'm testing this strange behavior using livevoip, teliax, and
voicepulse connect. I'm calling our office phone which picks up after
two rings and plays a greeting. With livevoip and teliax I hear 3-4
rings and when the line answers I
I have one with 33. but I can't get the voicemail to copy to more than
20 mailboxes.
-
Chris Stinson
Network Operations Center
ISDN-Net, Inc.
615-221-4200 x103
[EMAIL PROTECTED]
Eric Wieling aka ManxPower wrote:
Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?
Duane Cox wrote:
Do you get 2-way audio that sometimes drops off to 1-way audio then picks
back up as 2-way? (Thats what I see)
Not sure if my problem is a lost packet issue as I am sending IAX off net.
Duane Cox
- Original Message -
From: [EMAIL PROTECTED]
To:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Matthew,
Matthew Boehm escreveu:
| Rodrigo P. Telles wrote:
|
|
|Does someone knows if the next release of Asterisk (1.0.8?) will have
|Realtime support and when we will have the next Asterisk release
|with Realtime features?
|
|
| Where is your
Yep, seeing the exact same problem here if it's a trunked IAX2
connection. A CVS checkout I had from early April did the same thing.
Try setting trunk=no and see if it works. Seemed to fix the problem
here for us with our development cluster.
To quote bkw (from earlier this week in IRC),
How do I remove it from kudzu?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Robert Webb
|Sent: Viernes, 29 de Abril de 2005 08:57 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] Problems with
Thanks Daniel,
We may end up replicating your tests in order to confirm some of your
results. I don't know if it will be anytime soon, because we don't have
the hardware yet. Regardless, I will share my results with the list.
Anyone out there have any ideas on why the NFS mount affected call
Hi,
I am doing some testing with asterisk using Cisco IP Phones 7960's and
EyeBeam. I have canreinvite=yes on all my devices but the media still
goes through the asterisk box. Does it mean that Cisco and Xten do not
support re-invites? If yes can you recommend SIP phones or adapters
that
Does anyone have experience with using NAS
(http://en.wikipedia.org/wiki/Network-attached_storage) or SAN
(http://en.wikipedia.org/wiki/Storage_area_network) for this application?
Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian
Daniel Salama wrote:
Sure.
I
I don't have any failure I just want to know if the next release will
be 1.0.8 or 1.2.
Oh good grief. My fault..for some reason I read the subject and
processed failure instead of 'feature'. Thank goodness its friday.
Anyway, there will probably be a 1.0.8 release. But remember that 1.0.*
is it possible to program an adtran 600 to act as
the network and asterisk to be cpe?
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Callum,
Matt, is this similar to the idea that you have for your project ?
Similar, except we are looking to have a single Asterisk server attached
to the Gateway for centralized queuing, reportings, call recoring, etc.
We are a call center, so having everything in a single environment is a
Hello,
I am in search for a SIP or IAX softphone that works with * and
supports commercial codecs like g729 and g723.1. It can be commercial
license . I have been through Xten and SJphone.
Let me know anyone can offer this. I need it on an urgent basis.
Thanks.
Ehsanul Karim
Maybe something like this would be good.
http://www.pcmicrostore.com/PartDetail.aspx?q=p:10502197
- Original Message -
From: Matt Roth [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005 2:11 PM
-BEGIN PGP SIGNED MESSAGE-
Does anyone have any working example GR-303 configurations for zaptel
and
zapata conf?
The information available on the wiki as well as in the sample
configurations just doesn't seem to be enough to bridge the gap for
me.
In Zaptel.conf,
Do you set up a
Hi Matt,
Does anyone have experience with using NAS
(http://en.wikipedia.org/wiki/Network-attached_storage) or SAN
(http://en.wikipedia.org/wiki/Storage_area_network) for this
application?
I've had our agent/queue recordings dumped both to local disk and SAN
(currently using local disk as
Matteo Brancaleoni wrote:
yes, some multiplexer allows that, but they're quite expensive
compared to another E1 card for asterisk.
I think you'll need at least 1k $$$ for a such splitter.
Matteo, would you have any reference for this 'mux/splitter' ?
I would guess it need to be smart enough to
Sander wrote:
I compiled the bristuff drivers and then I do
--
When doing lsmod I can see qozap is loaded with zaptel but no entry in
/proc/zaptel/
Did the compiling go correct?
What version of bristuff are you using? (latest? 0.2.0rc8a)
What linux distro are you
List members,
Does anyone have an interest in forming a hardware architecture group?
It seems that Asterisk is so tightly linked to specialized hardware and its
corresponding architecture that developing the software alone is insufficient
for its adoption to large scale applications.
Thank you,
Andrew Kohlsmith wrote:
No; if the driver didn't load that's a major problem. Remember that if the
channel doesn't exist all the subsequent channels move up... serious
potential security issues.
Good points. What if it kept the number (so nothing moved up), but
marked the channel inuse (or
Which card do you recommend using instead of the tdm400p?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|John Novack
|Sent: Viernes, 29 de Abril de 2005 09:19 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re:
SIPp is a free Open Source test tool / traffic generator for the SIP protocol
http://sipp.sourceforge.net/
On Fri, 2005-04-29 at 14:14 +0200, Nils Ohlmeier wrote:
The homepage http://sipsak.org contains some examples. If you need help with
special cases drop me a line.
Regards
Nils Ohlmeier
On
I think that would be a great idea. The only problem I see is that
Asterisk is growing its feature set and maturing at such a dynamic
rate, that I don't know in many cases, where to point the finger at.
Sometimes it's stability of the CVS version, sometimes it's stability
of Digium or
Thx Rene, Ill give it a try
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|René Mayorga
|Sent: Viernes, 29 de Abril de 2005 12:29 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Traffic Testing
|
Any url?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|René Mayorga
|Sent: Viernes, 29 de Abril de 2005 12:29 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Traffic Testing
|
|I'm using sip-tester
I just called this company. They seem to do what is required. Now remains the
pricing part of it. I will wait for their feedback.
http://www.megatelindustries.com/products.htm
Hakem,
Selon Julio Arruda [EMAIL PROTECTED]:
Matteo Brancaleoni wrote:
yes, some multiplexer allows that, but
This is an interesting question. I haven't tested it but would love to
know if it works or not. Anyone?
- Daniel
On Apr 29, 2005, at 3:38 AM, Michael Welter wrote:
I haven't seen this before--can an agent log into a queue on a remote
(i.e. over IAX) Asterisk server?
Is there any way to detect * deadlocks automatically?
i.e with a running program in background.
Paradise Dove
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,n,Hangup
If I asterisk -r, when I dial the 888, I see
Userevent appearing in the
console.
However, if I telnet to the * manager using a name
and password that has
the user option, that telnet session sees
everything
Many channel banks have two T-1 connectors and support a feature called
'drop and insert'. This allows some of the DS0 channels to be cross
connected from one T-1 connection to the other. The first T-1 connection
can go to the telco or an interface card in a computer, and the second T-1
can
Hi,
I've been playing around with CFIM
and CFBS and came across something rather odd. I found that a SIP X-lite phone
didn't give the expected results when running the sample CFIM/CFBS code from
the Wiki - see
This is not quite on-topic for the Asterisk list, but is a much
higher chance that I will find a rich network of possible candidates
on this list than any other. Besides, with the amount of problems
that we all have with SIP and various CPE working with Asterisk, the
benefits of any
[EMAIL PROTECTED] 1.0 released
This is the first production release of [EMAIL PROTECTED]
We have worked hard over the past few months to make
[EMAIL PROTECTED] easy to use and stable. Thanks for all
the help with testing and fixes from [EMAIL PROTECTED]
users all over the world.
There are no
-Original Message-
From: Anton Krall [mailto:[EMAIL PROTECTED]
Sent: Friday, April 29, 2005 1:50 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Problems with TDM400P card
How do I remove it from kudzu?
I am
I seem to get bounces on DTMF.
For instance, if I turn on debug, and I dial the voicemail, and enter 1234
as extension, I see in the logs 12234 111234 12344 and so on, same
with passwords.
But dialing an extension never seem to fail this way.. Any hints?
smime.p7s
Description: S/MIME
On 4/29/05, Daniel Salama [EMAIL PROTECTED] wrote:
I think that would be a great idea. The only problem I see is that
Asterisk is growing its feature set and maturing at such a dynamic
rate, that I don't know in many cases, where to point the finger at.
Sometimes it's stability of the CVS
On April 29, 2005 02:54 pm, Jeb Campbell wrote:
I agree that it should be a very loud error (and possibly repeated
notifications on the console). But I also think that it should be able
to limp along. What would you think of a commercial phone system that
completely dies when one port dies?
Ok,
So I am trying to still figure out my ringing issues. This time I
grabbed the butt set I own and hooked it into my pots line. With the
butt set in monitor mode, I called the pots line so I could actual hear
the AC ring. It was a low frequency ringing sound like I am accustomed
to.
I then
Hi all,
I'm trying to use spandsp and asterisk to send faxes. To do so I am
creating tiffs with Ghostscript. When I use Ghostscript 6.50 it seems
to work fine, but when I create the tiff using Ghostscript 8.51 (or
7.06) txfax garbles the tiff and it comes through all messed up.
First of all is
Does anyone have any experience with servers from siliconmechanics.com?
Are they reliable? How does * run on them?
Thanks
- Daniel
On Apr 29, 2005, at 4:22 PM, snacktime wrote:
Personally I would buy an * box from someone like asaservers.com. At
least companies like that really know their
I would also be interested in alternatives to the Tdm400p. I have had endless
problems with a tdm400p card not being able to get the zttest numbers above
99.975 and as a result not being able eliminate an intermitent but consistent
echo.I have tried to date 4 different motherboard and hardware
I've seen this happen once or twice before. Both times, different
things fixed it.
On one of them, we tweaked on the echo canceller settings, and on the
other, I believe we tweaked on the rxgain/txgain settings.
On 4/29/05, Jan Johansson [EMAIL PROTECTED] wrote:
I seem to get bounces on DTMF.
Anyone use these with *? I'm curious to know how they compare to the
Hitachi WIP-5000?
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
Hello all,
We are considering implementing a new system based on Asterisk on the
back end. I am very intrigued by the IP phones, but I have two
questions regarding paging and intercom functions.
I know that * supports these functions, but I'm not sure I fully
understand how. On our existing
Jacob, all of these questions have been answered numerous times before,
please search the archives.
BTW the cheapest way to set up a fxs paging is by modifying a
grandstream bt101 with auto answer per zone.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
Polycom phones and Snom phones supoprt paging.
As far as your Overhead paging all you need is an FXO port on your
system. The * system will work perfectly with this. Even allowing the
zones to be set from the dialplan so your users won't need to learn any
new 'paging codes'
Email me off -list
I gave up with sipgate after dtmp tone recognition didnt work - and
found other who also have this problem and emails to sipgate are
ignored..
Rafal Kaniewski
-Original Message-
From: [EMAIL PROTECTED]
- Original Message -
From: Jan Johansson [EMAIL PROTECTED]
I seem to get bounces on DTMF.
For instance, if I turn on debug, and I dial the voicemail, and enter 1234
as extension, I see in the logs 12234 111234 12344 and so on, same
with passwords.
What type of phone SIP or analog? What
- Original Message -
From: Jacob Cazzell [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005 4:21 PM
Subject: [Asterisk-Users] Paging and intercom
On our existing phone system, if you dial an
extention the other end will beep and then setup an intercom
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a list
of callers to be blocked. When they call, they should hear busy and
then we hang up. We have about 100 DIDs routed to different contexts
and I wouldn't want to have to
I too wish I had a solution.
What I REALLY wish is that Digium would acknowledge that there is a
whole bunch of problems, firstly with the card and MANY motherboards,
then with reported problems some have with the FXO, either card or
drivers? and FXS problems as well, again with the card and
Hi,
I'm finding long timeout before DISA really calls extension user entered
annoying. I wonder what workarounds are you using for this ?
Playtones is one possibility , but it won't stop when user starts entering
numbers...
Regards,
Rob.
___
What are you using instead of SIPGATE in the UK ?
I also have this problem with DTMF tones not being passed to Asterisk from a
PSTN line and my e-mails are being ignored too !
If only they sorted that problem out, it would be a great service.
Thanks, Paul.
-Original Message-
From:
At 4:57 PM -0400 on 4/29/05, Daniel Salama wrote:
On Apr 29, 2005, at 4:22 PM, snacktime wrote:
Personally I would buy an * box from someone like asaservers.com. At
least companies like that really know their hardware, and if you tell
them the common issues with * they could probably put together
Sounds like a good idea to me. I would watch it.
Race Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
Sent: Friday, April 29, 2005 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
Hi,
I need some info from people with the x100p card (digium or clone),
please send me the output of lspci and lspci -n from your linux
machine, i am tring to find out something on my * server.
Thanks.
Marco.
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00:0e.0 Communication controller: Individual Computers - Jens Schoenfeld
Intel 537
00:0e.0 Class 0780: e159:0001
On Fri, 2005-04-29 at 16:26, Marco Supino wrote:
Hi,
I need some info from people with the x100p card (digium or clone),
please send me the output of lspci and lspci -n from
Marco, I've got a clone. X101P I think it was sold to me as.
$ lspci
...
00:08.0 Communication controller: Tiger Jet Network Inc. Intel 537
...
$ lspci -n
...
00:08.0 Class 0780: e159:0001
...
Mojo
Marco Supino wrote:
Hi,
I need some info from people with the x100p card (digium or clone),
please
You mention the WIP-5000, Does that handset have the ability to receive
text messaging/instant messaging?
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005
Hello:
I have searched everywhere in this list but cannot find the
.cfg file (ipmid.cfg) entry to set the initial ringer volume for an IP500.
Could someone please post the XML attribute and value to set
the ringer value, to say its maximum upon the phones restart.
THANKS IN
Daniel Salama wrote:
Question: how can I block someone from calling us?
Sometimes we get crank calls into our office. We'd like to build a list
of callers to be blocked. When they call, they should hear busy and then
we hang up. We have about 100 DIDs routed to different contexts and I
wouldn't
Clone here as well.
:00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
- DAn
Marco Supino wrote:
Hi,
I need some info from people with the x100p card (digium or clone),
please send me the output of lspci and lspci -n from your linux
machine, i am
Hey Mojo, I'm thinking you might try using priorty 's to set some kind
routing. just a thought..
Mojo Jojo wrote:
We recently had our PRI installed, we currently have 100 toll-free's
pointing to it.
I have almost everything working great but..
I have setup the first few numbers we want to use
Hi all,
Can someone point me in the right direction to configuring sendmail to work
with Asterisk voicemail and faxes?
I did a bit of research on the web but came up more confused that when I
started.
It's the basic setup I'm having trouble with, where to add the SMTP and
login and user name
I have [EMAIL PROTECTED] 0.9 running, and everything seems
to work well EXCEPT incoming calls.
I have an FWD and Teliax trunk (both using IAX), and a
Cisco 7960 SIP phone connected to Asterisk.
Everything tests fine:
- Can call from softphone to Cisco and vice versa
- Asterisk
From the CLI if you do a iax2 show registry, does it show you registered?
Maybe you can post the parts of your config that pertains to your question?
- Original Message -
From: Patrick Gray, Jr. [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 29, 2005 11:03 PM
Since all the asterisk program needs to do is send mail through smtp,
and since using sendmail for this purpose is a bit like using Jeff
Gordon's racing engine on a bicycle we opted to scrap sendmail and use
msmtp. This is basically just an smtp engine. To our mail server, it
looks just like any
Are you sure it's registering?
-
Dan Levine
CYTEXONE | Your Technology Specialists
t: 877.CYTEXONE x 810
l: 212.477.0990 x 810
e: [EMAIL PROTECTED]
http://www.cytexone.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Patrick
Gray, Jr.
Sent:
What's what I'm trying to avoid. To answer your question: I have TE4XXP
with T1s (not PRIs). What I want to do is block it based on the
caller-id and not the DID Number. That way, I don't have to write 100+
lines.
Thanks,
Daniel
On Apr 29, 2005, at 6:23 PM, Stefan Gofferje wrote:
Daniel Salama
Hi All
I am using asterisk to redirect some extension calls to few cell phones.
I was wondering if it is possible to have * display on the cell phone as 'PRIVATE NUMEBR' or 'CALLS' instead of the calling number.
Thank You__Do You Yahoo!?Tired of
I have two asterisk boxes connected using IAX. There are two T1s on
each box. I have all my dialing rules in one of the asterisk boxes and
all of my agents register on the same box where I have all the dialing
rules. See diagram below:
Asterisk_1 --2xT1-- PSTN
||
||
Asterisk_2 --2xT1-- PSTN
||
I'm working with SER + Asterisk. I was told that to have SER push calls to
multiple Asterisk servers, I can use the LCR Module, I'll just give all
the Asterisk servers the same weight/price, and SER will randomly send
outbound requests to each Asterisk server. It's not truly equally
balanced, so
Tim,
This certainly looks interesting. I just have a question about the
recipe: it makes reference to some AGI perl scripts. Is the source
available? Or may be it's irrelevant.
Thanks,
Daniel
On Apr 29, 2005, at 9:10 PM, Tim Litwiller wrote:
Daniel Salama wrote:
Question: how can I block
Bill Ford wrote:
Since all the asterisk program needs to do is send mail through smtp,
and since using sendmail for this purpose is a bit like using Jeff
Gordon's racing engine on a bicycle we opted to scrap sendmail and use
msmtp. This is basically just an smtp engine. To our mail server, it
On 04/30/05 02:42 Matt Roth said the following:
Does anyone have an interest in forming a hardware architecture group?
absolutely !
It seems that Asterisk is so tightly linked to specialized hardware and
its corresponding architecture that developing the software alone is
insufficient for its
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