On Sat, 30 Apr 2005, Ma Zhiyong wrote:
> I use TE405P as gateway and Eicon PRI card as fax card.
>
> When I receive the caller number from PSTN, I found it was 51863500. While I
> dial the FAX trunk, FaxGetty get the caller number 051863500.
>
> -- Executing NoOp("Zap/124-1", "51863500") in new
This question will be better addressed on the aah forums.
I would suggest:
1) have you setup a DID?
2) take a look in the log file
tail -f /var/log/asterisk/full
3) see the numerous threads on the aah forums about how to configure FWD
and Teliax (and other providers)
I personally have both FWD
Hello,
from what I see, I guess they're only ways to insert a piece of speech
without recording it; you could easily record the phrases yourself and add
Playback()s instead.
BTW, I'd like to thank Tim for sharing his recipe with us. Anybody else's
got a recipe to share? :-)
l.
In data Sat,
Hi, all. I'm glad I put asterisk and hylafax
together just like PSTN->Asterisk->Hylafax->Email. And the
fax2email function works well.
But I also find some bugs about CID number.
I use TE405P as gateway and Eicon PRI card as fax
card.
When I receive the caller number from PSTN,
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 ad
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
looks
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 someo
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 on
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
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
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
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:
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 ot
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:
Sent: Friday, April 29, 2005 11:03 PM
Subject: [Asterisk-Users] Can
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 inboun
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
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
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
> machi
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
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 phone’s restart.
THANKS I
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"
Sent: Friday, April 29, 2005 4:14 PM
Subject: [Asteris
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),
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 -
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.
___
Asterisk-Users mailing list
Asteris
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] Asteris
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
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: [EM
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.
___
Asterisk-
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 dri
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 manua
- Original Message -
From: "Jacob Cazzell" <[EMAIL PROTECTED]>
To:
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 channel
that's hands free for t
- 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
I gave up with sipgate after dtmp tone recognition didn’t work - and
found other who also have this problem and emails to sipgate are
ignored..
Rafal Kaniewski
>-Original Message-
>From: [EMAIL PROTECTED] [mailto:as
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 of
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] [mailto:asterisk-use
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 pho
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]
o713-
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 DTM
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 c
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 hardwa
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 th
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 hoo
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 die
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
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 cr
> -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?
>
[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 new
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 improvemen
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
http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding
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 g
http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Events
,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
> ev
Is there any way to detect * deadlocks automatically?
i.e with a running program in background.
Paradise Dove
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
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?
_
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
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
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
|
|I'
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 whose
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 Fr
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: [Asterisk
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 (o
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,
M
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 runnin
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 dig
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
-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 GR-
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"
Sent: Friday, April 29, 2005 2:11 PM
Subject: Re: [Asterisk-Users] As
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
__
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
h
is it possible to program an adtran 600 to act as
the network and asterisk to be cpe?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
ht
> 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
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
> 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
>
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
q
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 TDM40
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), "trunk
-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 fail
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:
Sent: Friday, April 29, 2
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?
___
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 an
I'm using "sip-tester" you should try it
gnuws:~# apt-cache search sip-tester
sip-tester - a performance testing tool for the SIP protocol
gnuws:~#
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 ca
Guy Boehm wrote:
wau thank you it works!! but,
first it says that e loop is detected,
and secondary what must I do to hand over the new working channel to
my x-lite to use it???
DENGENS
Richard Lyman <[EMAIL PROTECTED]> wrote:
Guy Boehm wrote:
> fputs($socket, "Channel: 6159bfb47
David Josephson,
Not off-base, but you haven't made it all the way home yet. This is
another layer of the puzzle, and again we are not talking about apples
and apples here. "Circuit switched" means that there is a (real or
virtual) circuit that takes data on an input port and delivers it to
an
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:
Sent: Friday, April 29, 2005 12:03 PM
S
Daniel,
Thanks alot for this post. You were right on time with valuable
information.
Thanks again,
Steve
- Original Message -
From: "Daniel Salama" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, April 29, 2005 12:37 PM
Subject: Re: [Ast
There isn't a specific command in the manager API itself to do it.
However there is a CLI command and you can use the manager command
action to get the information. Below is an example, you will need to
parse the response part to see who is connected.
Action: Command
Command: show manager con
Ok, this is probably stupid question of the week. I have
exten => 888,1,whatever
exten => 888,n,UserEvent(Event|Data)
exten => 888,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
th
Upgraded one of my asterisk servers to the latest cvs head version last
nigh now I get one way audio on IAX2 channels when calling other
asterisk servers. Anyone seeing the some problems?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
h
On April 29, 2005 12:38 pm, Jeb Campbell wrote:
> While I like the idea (and will look into it -- might need a wait, etc),
> as I said in original post, unloading and reloading did not fix the
> problem. It took a clean shutdown (unload and restart) to fix the problem.
Hmm; that is odd...
> So r
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 support r
Greetings,
I have two machines. One is a P3 Dell Dimension 4100, the other is a
PowerEdge SC420. Both are running Asterisk 1.0.7, the PowerEdge has a
TE405P card in it, the Dimension has a Digium X100P present (although
not modprobed). Each machine has mpg123 0.59r loaded, and is using
the exac
Well, I don't think I'm ready to spend that much money :)
I understand your point regarding that load depends on usage.
SIP_Agents are simply agents answering calls. Average call length would
be about 8 minutes. During some of these calls (maybe 25%), agents will
conference the call (PSTN channe
Andrew Kohlsmith wrote:
It has nothing to do with not being unloaded; I've seen the wctdm driver fail
to detect modules correctly. Run it again and it works just fine. Some kind
of minor tweak is in order, I believe.
As an interim solution, your asterisk starup script should try to unload any
Sure.
I setup a small lab on a machine with 4 T1s and 36 agents logged in.
The system was configured to Monitor all outbound calls as well as
monitor all calls distributed by Queue app (monitor-format setting in
queues.conf).
When recording to local disk, everything was working fine. Agents
Wouldn't introducing Samba into the mix be even worse?
I would think it would add more processing power and network use to be
constantly writing over the network as opposed to recording on the
same box.
If it's such a critical system, it should have the power to do that,
but that's not the point.
On Friday 29 April 2005 12:12 pm, Kib Eki wrote:
> Hi,
>
> when I dial my voicemenu the menu voice is always cutted so that i only
> hear 'word from password.
> What do i have to configure so that i hear the full text from the
> beginning?
>
> thanks, Kib
You might try inserting a Wait in your me
Hi,
when I dial my voicemenu the menu voice is always cutted so that i only
hear 'word from password.
What do i have to configure so that i hear the full text from the beginning?
thanks, Kib
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.c
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.
Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto:
> Hi,
>
> Assume I have one E1 digium card to which
I am running on usermodelinux
Itamar Reis Peixoto
+55 (34) 3238 3845
e-mail : [EMAIL PROTECTED]
http://vps.ispbrasil.com.br --->>> servidores linux
Has anyone had any experience (good or bad) running Asterisk under
VMWare ESX server on a blade chassis? This application will (fairly
obviously) no
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 failure? I don't see anything. The next stable release of
asterisk will be 1.2 and it will ha
On April 29, 2005 11:22 am, Jeb Campbell wrote:
> As soon as power came back, the server started. However when it loaded
> wcfxs, port 3 on the card failed the tests (I assume from the module not
> being unloaded before power off). Because this one port failed the
> test, chan_zap failed to load
1 - 100 of 176 matches
Mail list logo
