2010/8/13 Lyle McKarns lyle.mcka...@nexusmgmt.com:
Does anyone have any feelings one way or the other about running Asterisk on
AMD vs running Asterisk on Intel?
Only political feelings. I want to support AMD so there's at least
some token competition for Intel.
Both companies make nice 6-core
On Wed, Aug 11, 2010 at 10:12 AM, Kent Varmedal k...@domeneshop.no wrote:
I'm trying to set up an old PBX (that supports SIP) to go through our
new Asterisk server, so that our old phones can be used still for some
time.
How can I set up Asterisk to deliver a trunk sip connection that our old
On Wed, Aug 11, 2010 at 4:36 AM, Tino t...@sparksupport.com wrote:
Is it possible to install Asterisk on Vmware(centos) from source. Is there
any difference or disadvantage for this compared to asterisk running on
physical machine.
This has come up repeatedly on the list.
Basically, the less
On Wed, Aug 11, 2010 at 11:24 AM, Kent Varmedal k...@domeneshop.no wrote:
We need to upgrade this PBX for it to work with SIP, it is at the moment
using ISDN. And those who delivered it and do the
support/reconfiguration is paid by the hour. We don't have any control
over it our self, so when
On Mon, Aug 9, 2010 at 4:36 AM, Zhang Shukun bit...@gmail.com wrote:
hi, group
there are two module can used for meeting. MeetMe and
Conference(which is a plugin)
My question is :
which is better for large conference(maybe above 100 people in a meeting)?
There's at least one more
On Wed, Aug 4, 2010 at 11:57 AM, Alejandro Cabrera Obed
aco1...@gmail.com wrote:
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
Not really an asterisk
On Thu, Jul 29, 2010 at 4:15 AM, zeynep yildirim zyildi...@gmail.com wrote:
Hi All,
I upgraded 7970 from SCCP to SIP. But the phone isn't registering.
Have you got any working XML file for 7970 phones.
Isn't registering with what?
If you're registering that with CallManager, you have to
On Thu, Jul 29, 2010 at 7:22 AM, Nguyen Quang Tri kihote...@gmail.com wrote:
Hello,
i have Cisco Unified Communications Manager with 10 ip phone,i dont buy
license IVR of Cisco Unified Communications Manager. Can i use feature IVR
on Asterisk connect with Cisco Unified Communications Manager.
So historically I've done one of two things on systems where I've
needed to use MeetMe
* used a real Digium card, and I've only ever used a TE400 or a TE420
for that purpose, and I know they have the timing chip
* used dahdi_dummy, which works well with light load, but I had it
running on a very
On Thu, Jul 29, 2010 at 5:04 PM, unsero...@aol.com wrote:
Do you know if it is possible to interconnect 1.6 with Microsoft Office
Communications Server 2007 and use the Office
Communicator as a softclient for telephone calls and the Communicator for
Instant Messaging? I believe you can set up
On Tue, Jul 27, 2010 at 6:08 PM, bruce bruce bruceb...@gmail.com wrote:
:-) I knew someone would bring up FreePBX. I have FreePBX installed and it's
not good for Queues at all. It's using the reporting tool from Areski and
One of the several things you asked for was GUI for cdr database logs.
On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce bruceb...@gmail.com wrote:
I seem to not be able to find any good open source Asterisk Queue Analyzer
and Asterisk Log Analyzer on the web.
google 'freepbx'
It does some of what you want. For the rest of what you want, strongly
consider paying a
The only big difference I know, is:
VicidialNow - *based on CentOS* - Vicidial 2.0.5.1rc1
ViciBox - *Based on OpenSuse* - Vicidial 2.0.5
The core of the call center for both of them is Vicidial.
Regards.
2010/7/25 Alejandro Cabrera Obed aco1...@gmail.com
Dear all, I need a call center
On Fri, Jul 23, 2010 at 8:46 AM, Matt mhop...@gmail.com wrote:
It's not necessarily this simple. There is an approximately 50-75foot cable
run through ceilings and walls (CAT5) to the location where the phones will
be. At the phone location there is no power.
You always have options. You
There is no such device -- it's outside of the POE spec.
Class 3 devices are allowed to consume at max 15.4W. Most phones are class 3
devices. The math just doesn't work out. Even if you used the draft standard
for class 4 (~30W), you could still power max 2 devices at 15W/ea.
-Dave
On Thu, Jul
On Thu, Jul 22, 2010 at 2:46 PM, Matt mhop...@gmail.com wrote:
I've got an interesting situation where I have one cable run from the feed
area to the service area. I have three devices that I need to power at the
service area. Is anyone aware of a device that will take the POE from the
On Sat, Jul 17, 2010 at 6:52 PM, David Shauger sollost...@gmail.com wrote:
Can anyone provide the settings in Audacity to create a proper wav file
without having to do additional conversion in the cli? Has to be a way to do
this with less steps.
If your goal is to 'minimize steps', you should
Can anyone provide the settings in Audacity to create a proper wav file without
having to do additional conversion in the cli? Has to be a way to do this with
less steps.
David Shauger
Vice President
Sollos Technology
should I check other configuration choices?
Any help would be appreciated :)
Thanks in advance,
David
--
C/JosEchegarayn8
Edificio3,1Planta,mdulo1
ParqueempresarialAlvia
28230LasRozas
Madrid,Espaa
DAVIDMATASHERNNDEZ
configuration choices?
Any help would be appreciated :)
Thanks in advance,
David
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New to Asterisk? Join us for a live introductory webinar every Thurs
On Sat, Jun 26, 2010 at 2:09 PM, Eyal Goltzman egoltz...@gmail.com wrote:
Hello,
After installing and learning Asterisk I found myself with a need for a
minimal set of empty configuration files with only the must have stuff in
order to setup a SIP only machine, is there a place to find it?
On Thu, Jun 24, 2010 at 11:24 PM, Cary Fitch ca...@usawide.net wrote:
But, we have an opportunity to get into a big time telecom activity.
It would have 2000 to 30,000 user lines per city, and we would like to have
those brought back to a central location for control and because transport
can
On Fri, Jun 25, 2010 at 11:00 AM, Cary Fitch ca...@usawide.net wrote:
I see some talking about TNTs in this forum. Those are 672 lines or in some
versions double that, what is used behind them to do the processing, etc.
So a channelized DS3 is roughly 28*23 channels in US if you do one
On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir nde...@demir.web.tr wrote:
This is a really rookie question: when should i use TE110P ISDN PRI Card?
--
Necati DEMİR
When you have a single PRI / BRI line you wish to terminate into an
asterisk system.
--
configuration choices?
Any help would be appreciated :)
Thanks in advance,
David
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Hello List:
I'm working on a funny scenario, where I'm bouncing calls from a Cisco
call center into asterisk. Cisco call center has some logic that if a
customer calls in, an agent is logged into a given extension... if
Cisco sends a customer call to that extension, and there is a ring
with no
Your installation should work, you must configure the card channels and
load the card module on your OS.
Regards.
2010/6/16 Alejandro Cabrera Obed aco1...@gmail.com
Dear all, I have to install an E1 card in my Asterisk 1.4.23 server
and here is my short question:
Is it necessary to install
On Wed, Jun 16, 2010 at 11:50 AM, Tilghman Lesher tles...@digium.com wrote:
On Wednesday 16 June 2010 08:21:17 David Backeberg wrote:
I know if I do not do an Answer() that the call is not yet picked up.
However, if I do a HangUp(), is that functionally equivalent? Can you
Hangup() a channel
We have two models of Cisco phone that do this, but Asterisk handles it fine.
I don't recall doing anything special to make it work. We're using Asterisk
v1.4.17.
What is the error message on Asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of
On Sun, Jun 13, 2010 at 2:59 PM, Vieri rentor...@yahoo.com wrote:
I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6
compatible).
I've taken a look at CAGI and QUIVR but their latest code releases date back
to 2006.
I've also seen a more recent project (wildpbx)
On Thu, Jun 3, 2010 at 3:56 PM, cov...@ccs.covici.com wrote:
Hi. For several months now asterisk will mysteriously stop inserting
records into cdr database. I am using mysql and the asterisk addons
1.6.2 to accomplish this. Sometimes there is a strange error about
column names, but often
On Thu, May 27, 2010 at 6:17 PM, Theo Band theo.b...@greenpeak.com wrote:
I used to build Asterisk from source including the zaptel-dummy module.
Last year I decided to upgrade and use a yum repository. I hoped that
this would be less hassle compared to manually chasing after the latest
On Mon, May 24, 2010 at 9:41 AM, Kingsley Tart kings...@skymarket.co.uk wrote:
I know nothing of Trixbox but I had a problem with my own dialplan where
there was a delay with the user selecting 0 from my IVR menu. It turned
out that because my extensions all started with 0 (they were real phone
On Thu, May 27, 2010 at 4:05 AM, Theo Band theo.b...@greenpeak.com wrote:
First I noted that dahdi_dummy is no longer present in
kmod-dahdi-linux-2.3.0.1-1.
Not exactly true.
myhost01 asterisk # lsmod | grep dahdi
dahdi_dummy 5812 0
dahdi_transcode 8968 1 wctc4xxp
On Mon, May 24, 2010 at 7:31 AM, Marcin J. Kowalczyk
marcin.kowalc...@ccig.pl wrote:
Medium load system (~300 simultaneous calls) crases few times a day.
1.6.1.19 but then upgraded to 1.6.2.7 but it's not solving issue.
Any idea what can be wrong/tunned?
I've three times had unexplained
Leif - thank you! Will try that.
On Fri, May 21, 2010 at 12:19 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
David Cunningham wrote:
Hello,
We're seeing lots of warnings like the following, running Asterisk
1.6.1.12. Does anyone know the cause or cure?
One explanation I've come
On Thu, May 20, 2010 at 11:41 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I'm evaluating what could keep me from upgrading production systems to
1.6.2.
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Have you met
I don't see anything in the SIP trace related to the warning messages.
Would anyone have any further tips?
Thanks for any help!
On Wed, May 19, 2010 at 9:12 PM, David Cunningham
dcunning...@voisonics.com wrote:
What should I expect see if it is the peer asking us to slow down RTP?
Thanks
] WARNING[27482] channel.c: Exceptionally long voice
queue length queuing to Local/12126412...@asterisk-phone-7e3d;1
Thanks in advance!
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Wednesday, May 19, 2010 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cause and cure for Exceptionally long voice
queuelength queuing to Local?
Hello,
We're seeing lots of warnings like the following
On Wed, May 19, 2010 at 12:38 AM, Adolphe Cher-Aime achera...@gmail.com wrote:
Hello Everyone,
I must deploy an asterisk system that can support
at least 500 pstn outbound calls.
It's a challenge as it's the first time i'm gonna build such a large
system.
I want to
On Tue, May 18, 2010 at 6:14 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
I read on voip-info.org that Asterisk 1.4 support T38 passthrough.
That may or may not be true. I do not know.
I do know that I've had much better success with fax in 1.6 than I
ever had in 1.4.
My personal
Hi Kevin,
We don't have mohinterpret set at all, so I think it uses default.
Is there anything else you can suggest? Any other places to go for
help?
Thanks for your assistance!
On Thu, May 13, 2010 at 11:32 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/13/2010 05:16 PM, David
Hello,
If you have canreinvite=no and a peer sends you a re-invite, what will
Asterisk reply with?
Thanks,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
have any idea why this is, or where I could go for more information?
Thanks for the help.
On Thu, May 13, 2010 at 11:06 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/13/2010 01:41 PM, David Cunningham wrote:
If you have canreinvite=no and a peer sends you a re-invite, what
-15.
a=sendrecv.
a=ptime:20.
Any help would be much appreciated!
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
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-- Bandwidth
On Wed, May 12, 2010 at 7:45 AM, Steve Underwood ste...@coppice.org wrote:
On 05/12/2010 08:46 AM, David Backeberg wrote:
So buy an asterisk appliance that supports fax, and then you can pay
somebody else to do the upgrade.
Does that appliance actually support FAX? The web pages don't mention
On Wed, May 12, 2010 at 9:53 PM, Ben Dinnerville b...@voicelogic.com.au wrote:
We are still getting an issue with a particular file which I have tried
multiple different ways to create to no avail. The tiff file is created
with ghostscript from a pdf as per the guidlines but every time we try
Make it say 'zed'.
It will make the British happy, and cause a different kind of
confusion for the Americans.
On Tue, May 11, 2010 at 4:09 PM, Richard Kenner ken...@gnat.com wrote:
This one works on my box (Vestec on 1.4.30 on OpenSuse)
Hmm... Not for me.
$Digit = (ONE:1 |
TWO:2 |
THREE:3
Ummm, zed is z.
I was thinking of nought.
On Tue, May 11, 2010 at 8:39 PM, David Backeberg dbackeb...@gmail.com wrote:
Make it say 'zed'.
It will make the British happy, and cause a different kind of
confusion for the Americans.
On Tue, May 11, 2010 at 4:09 PM, Richard Kenner ken...@gnat.com
On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Anybody know a reliable fax solution for 1.4.30 branch?
I am using PikaFax on another server and works very well (about 3000 faxes
a week), but it appears they no longer offer their product
On Tue, May 11, 2010 at 11:32 AM, Carlo Dimaggio jaasmail...@gmail.com wrote:
Hi all,
In order to use the open url function of zoiper (it opens an url
based on the asterisk $callerid(dnid)), I need rewriting of the dnid.
In my dialplan I have:
exten = 1000,3,Set(CALLERID(dnid)=newdnid)
On Mon, May 10, 2010 at 7:19 PM, Richard Kenner ken...@gnat.com wrote:
Which speed recognition products will also recognize DTMF? In other words,
I want to say Please speak or dial the conference number. Does Vestec
allow that? LumenVox? Any other way?
You're on your own for making custom
On Sat, May 8, 2010 at 7:21 AM, Steve Totaro
stot...@first-notification.com wrote:
Maybe there is a simple setting somewhere, but RTFM from Digium tech
support when the FM offers no suggestion on how to possibly tweak settings
for better success.
Do you want to dump some samples from your
,
David
On Wed, May 5, 2010 at 6:53 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
I set: sip debug peer 3000 (my test extension) and dialed #3643873
Your X-Lite softphone actually calls %233643873 and not #3643873.
You would need to check the SIP RFCs in order
is
*also* a unique ip subnet (else how do all the vlans access a common default
gw?)
place the phones in a voice vlan, and the phone problem is solved.
as for the PC isolation, you might get better feedback on a cisco or other
networking forum.
-david
In case anybody was following this thread,
wanted to let people know that the fix made it into SVN,
and is packaged into
1.6.2.8-rc1
Huge thanks to Kevin and Tilghman
On Wed, Apr 21, 2010 at 3:40 PM, David Backeberg dbackeb...@gmail.com wrote:
issue opened.
https://issues.asterisk.org
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable. There
is no activity in the CLI. Other
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because
my other 2 1.4.30 boxes wouldn’t talk
-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 8:39 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other
-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 9:31 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
Nothing..goes directly to The person you are calling is unavailable.
On Wed
command in your CLI output.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 10:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 10:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Hash Dial Pattern Problems
I am on the 1.2 box
:
10.1.1.1:44392 - register (Via: SIP/2.0/UDP 10.1.1.1:5060;branch=) -
asterisk
asterisk should send a response back to 10.1.1.1:5060
from asterisk cli, run 'sip set debug' and post a copy of the REGISTER and
asterisk's response.
-David
-Original Message-
From: asterisk-users-boun
and therefore have no backward knowledge).
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel
*Sent:* Wednesday, May 05, 2010 4:41 PM
*To:* Asterisk Users Mailing List - Non
in the SIP/2.0 180 Ringing, the SDP shows:
a=sendonly
this is hold by rfc 3264. then when the other end picks up, a new SDP is
probably sent with
a=sendrecv
I believe your server is acting correctly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of
I don't know in your particular case, but if I call a PSTN endpoint via my
provider, the SIP signaling is different than if I'm calling a remote SIP
endpoint. This is because PSTN gateways have to make decisions (about codecs,
eg) independently of the remote endpoints.
In other words,
2010/4/29 garge rama garge.r...@gmail.com
Hi,
I am new to asterisk and trying to make calls with TDM400P asterisk digium
card.
I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and
libpri-1.4.10.2 packages which are downloaded from asterisk website (
www.asterisk.org)
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay. I'm
seeing prices around $45 to $50 for this thing. It comes with two FXS
ports, but I only need one
all you need to do is make the configurations mirror each other.
in the example below, all of the endpoints are SIP, but it doesn't matter if
you move the endpoints to another protocol, like Fxs:
on serverA
extesions.conf:
[phones]
include = localphones
include = to_serverB
[localphones]
On Mon, Apr 26, 2010 at 11:33 AM, Olivier oza_4...@yahoo.fr wrote:
2010/4/26 Olivier oza_4...@yahoo.fr
This Replaces header refers to RFC3891 which is not yet supported in
Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA)
I took a look at chan_sip.c and read this :
/*
On Sun, Apr 25, 2010 at 7:13 AM, russian qwerty
russian.qwe...@gmail.com wrote:
Hello, David.
Thank you for reply. But my problem is certainly in the size of JitterBuffer
of chan_local. I realy need to know how to change the size of JB (reduce).
BTW:
1. The file /etc/asterisk/dsp.conf doesn't
On Fri, Apr 23, 2010 at 3:21 PM, ad...@3a.hu wrote:
i have to put an * between two other SIP gateways and due to some
circumstances, i have to use sip over tcp. With 1.6.2.6 this is working
fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B
(ocs) and that's about it. In
On Fri, Apr 23, 2010 at 4:34 PM, russian qwerty
russian.qwe...@gmail.com wrote:
Hello.
As I see, there is a lot of threads about jitter buffer... Maybe anybody
knows something about my case? Any help will be appreciate.
So, the problem with voice quality was completely solved, BUT some
call-id doesn't match?
SIP/2.0 200 OK
...
Call-ID: 2117388659-506...@82.158.83.xxx
...
ACK sip:6615xx...@130.117.xxx.xxx SIP/2.0
...
Call-ID: 2117388659-506...@192.168.1.100
...
I'm not sure, but I think that the part after the '@' must also match
throughout the dialog. A Grandstream bug?
I didn't know there was an RPM for centos with asterisk in it.
I personally think that's a bad idea. There are a lot of source options.
app_fax.so in particular depends on SpanDSP, and particular versions thereof.
That's probably why it's missing from somebody's RPM.
Build from source.
On
On Thu, Apr 8, 2010 at 6:04 PM, David Backeberg dbackeb...@gmail.com wrote:
On Thu, Apr 8, 2010 at 5:01 PM, Kevin P. Fleming kpflem...@digium.com wrote:
David Backeberg wrote:
I'm doing really, really innocent things, like:
exten = s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
So I did some
On Thu, Apr 8, 2010 at 5:01 PM, Kevin P. Fleming kpflem...@digium.com wrote:
David Backeberg wrote:
I'm doing really, really innocent things, like:
exten = s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
So I did some more testing. Same dialplan, reverted to
asterisk-1.6.0.13, and the contexts
issue opened.
https://issues.asterisk.org/view.php?id=17223
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running on
other Dell servers, but this is a rack with Xeon processors. I can't see what
else might be different to cause this. SIP only, no cards. Running on CentOS
5.2.
David Shauger
Vice President
Sollos Technology
I use the option 'r' on 1.4, to record the meetme application. Asterisk
leaves these records at /var/lib/asterisk/sounds/meetmeXX.
take a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe if you
need more information.
Regards.
2010/4/14 Renato bianchini renato...@yahoo.com.br
Please note: A Zaptel timer must be present for conferencing to work!, but
if the user does not have ZAP/DAHDI hardware, he can use ZAP/DAHDI DUMMY
2010/4/15 Luki lugos...@gmail.com
I use the option 'r' on 1.4, to record the meetme application.
Asterisk leaves these records at
On Wed, Apr 14, 2010 at 4:55 PM, Stéphane Bauland baula...@epitech.net wrote:
I'm planning of creating a speech/video conference application. This
application will provide a system to see/listen to each personn present
in the conference.
Else, do you know any other way to do this ?
On Wed, Apr 14, 2010 at 8:55 PM, Edwin Quijada
listas_quij...@hotmail.com wrote:
My problem is that I need to execute windows app using IVR in Asterisk so we
What is the windows app that you cannot replace on Linux?
How about wrapping THAT program with simple inputs and outputs, and
build a
On Tue, Apr 13, 2010 at 12:56 PM, Ricardo Coelho
ricardo.tch...@gmail.com wrote:
Hi there,
Does asterisk keeps the master.csv open between writes? Right now I have 2
asterisk nodes sharing every configuration file (by using a distributed
filesystem) except the master.csv files. If asterisk
On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias ing.diasda...@gmail.com wrote:
What do you mean with problems on my configuration?
This is a FXO port on zapata:
signalling=fxs_ks
group=0
channel = 1
Not a FXS...can you explain to me what were you trying to say?
On Tue, Apr 13, 2010 at 6:55 PM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, Apr 13, 2010 at 4:12 PM, Danny Dias ing.diasda...@gmail.com wrote:
What do you mean with problems on my configuration?
This is a FXO port on zapata:
signalling=fxs_ks
group=0
channel = 1
Not a FXS...can
On Sun, Apr 11, 2010 at 5:00 PM, Danny Dias ing.diasda...@gmail.com wrote:
This digium card has 3 FXO ports and 1 FXS port where we have a fax
machine
connected!
The problem is that we can receive fax very good, but we can't make any
outbound fax call, in fact, our asterisk get freezed in
On Fri, Apr 9, 2010 at 5:40 PM, Joe Freeman j...@ngn-networks.com wrote:
I have res_fax setup and working for the most part. However, I'm seeing
some fax machines drop the connection on me -
Apr 9 17:33:11] NOTICE[30809]: res_fax.c:906 generic_fax_exec: Channel
'DAHDI/1-1' did not return a
On Fri, Apr 9, 2010 at 3:26 PM, Aldo Bergamini aabe...@nb-a.com wrote:
Hi all,
I have noticed something I can't solve regarding Asterisk (latest
1.6.0.x).
My server is set at the GMT+2 timezone. The clock is ok (I can get the
correct time at the terminal). But today I got a call at a time
On Mon, Apr 12, 2010 at 2:23 PM, Tim Nelson tnel...@rockbochs.com wrote:
- Vieri rentor...@yahoo.com wrote:
Hi,
What can I make of the following log messages? Extension 7114 tries to
reach 6035 but gets an unknown channel type. What does it mean?
(supposedly, 6035 was not busy...)
Apr
On Sat, 10 Apr 2010 22:34:28 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Just a heads-up ... my home asterisk server is being flooded by someone
from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it -
they're trying to send SIP subscribes to one account -
On Sun, 11 Apr 2010 08:09:02 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Look what they did to my latency, Gordon:-
http://f8lure.mouselike.org/archived_graphs/westek.bizorg.co.uk_day10.png
Oddly enough my latency wasn't being affected at all - however what I was
seeing
On Thu, Apr 8, 2010 at 10:33 PM, Alan Zheng machinecat1...@gmail.com wrote:
Hello All:
I saw there are app_fax and app_chanspy modules in 1.6.2.6, but there is NO
sample configure file for them.
Is anybody know how to use them, or where is the documentation for them?
If you read the code for
is connected to. The Polycom has 4 by RJ45 connections for the
512kbit/s service.
Any help would be appreciated.
Regards
David.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
On Wed, Apr 7, 2010 at 10:12 PM, Pham Quy qu...@vega.com.vn wrote:
Hi all,
I want to have a separate file to log what i need for my dialplan
without all output from Asterisk. By this way, i can easily to trace
problems caused by my dialplan.
How can i do that?
That's honestly a pretty
On Thu, Apr 8, 2010 at 4:30 PM, David Backeberg dbackeb...@gmail.com wrote:
However, something is really weird when I need to do System() calls.
It almost feels like delay in reading loopback, or running out of
available files on the system, or something like that. I'm rebooted
I've just upgraded to 1.6.2.6 on one of my test systems. I started out
happy, with some improvements in transfers to Local() channels from a
SIP channel, and much nicer verbose fax handling.
However, something is really weird when I need to do System() calls.
It was really, really weird. This was
On Thu, Apr 8, 2010 at 5:01 PM, Kevin P. Fleming kpflem...@digium.com wrote:
David Backeberg wrote:
I'm doing really, really innocent things, like:
exten = s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
So I did some more testing. Same dialplan, reverted to
asterisk-1.6.0.13, and the contexts
On Tue, Apr 6, 2010 at 12:36 AM, huu giang huugiang...@yahoo.com wrote:
Dear List,
Are there any way of configuring of Asterisk so it'll cache sound files in
memory, and when Asterisk receive a call, instead of loading sound files from
the disk, it will load from the memory and so Asterisk
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