ram ha scritto:
i have local extensions
and i have connected sip provider account to call out side
but i have account can call any part of the world
how to restrict some of users should call only USA or any Other
In a hundred of ways, I think the most straightforward is making a table
Hello,
Does anyone have some ideas on how to setup a shared line on several
SNOM phones in a reliable manner?
For the following scenario:
Calls enter on number 123. They do not have to ring anywhere (although
this could be the case), I just want the LEDs on a series of SNOM light up.
Phones
Hi,
I need to compile zaptel from asterisk 1.2.0 for linux kernel 2.4.26. I
make and make install it without any error, but I can't load ztdummy module:
# lsmod | grep z
Module Size Used by Tainted: P
# modprobe zaptel
# lsmod | grep z
Module Size Used by Tainted: P
zaptel 184384 0 (unused)
#
Thanks, your are absolutely right - I was thinking of REGISTER.
I couldn't find any information about that - if it's a known problem why
it's so hard to find any info?
Kevin P. Fleming napisał(a):
lokotes wrote:
When sip device sends to Asterisk INVITE with no 'Contact' field, the
server
Hi all,
Just got some eybeam xten pro and wooksung electronics sip
video phones.
Was testing it with a sip trunk to call manager. The voice
has no issues. But ofcourse the video doesnt go through the sip trunk
as I have no cisco video phones.
Managed to dial a xten pro extension
On Tue, Dec 06, 2005 at 11:34:51AM +0300, Eugene Prokopiev wrote:
Hi,
I need to compile zaptel from asterisk 1.2.0 for linux kernel 2.4.26. I
make and make install it without any error, but I can't load ztdummy module:
# lsmod | grep z
Module Size Used by Tainted: P
# modprobe zaptel
#
amaury BOSSE a écrit :
Thanks for your answer but I don't want to include a file, I only want to
include a variable.
Is it possible to execute linux commands like grep or top in a .conf file in
order to parse a file and get a variable?
AGI is your friend
-Message d'origine-
De
Second Post
!!! Please help !
Hi,
I'm using Asterisk
with a BRI Card (HFC Chipset) using the zaphfc driver.
I'm encountering the
following problem : when the first line is in use and a second incoming call
arrive, the console shows the following message :
Dec
I'm not using any sort of FXO adapter. I use a DID
from a voip provider as my dedicated DISA line.
- Original Message -
From:
AR
Tarzi
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, December 05, 2005 7:33
PM
Subject: Re:
Hi all, I have installed and configured asterisk on my debian machine. Right now i m making asterisk server for making connection between 2 X-Lite phones. I m working on different applications (voicemail, call queuing etc). I m plannning to take new hardware (digitnetwork's X100p FXO card) to
Hi all,
i run into problems using park calling with chan_capi.
My setup looks like this
[200X]--[Asterisk]--[PSTN]
For internal calls [1] and for incoming call from
PSTN[2] every thing works fine. Unfortunately when a
sip extension (say 2007) makes an outgoing call to
PSTN and 2007 tranfers to
On Tue, Dec 06, 2005 at 01:48:09AM -0800, Tejas Shah wrote:
Hi all,
I have installed and configured asterisk on my debian
machine. Right now i m making asterisk server for making connection between 2
X-Lite phones. I m working on different applications (voicemail, call
Title: Message
2
Things... You probably know anyway...
Early
echo cancel for satellite just used half duplex switching as a way to get around
the echo (This led to the echo suppess tone being used for faxes and modems to
allow full duplex).
From
my Telephony days the echo comes mostly
On Tue, Dec 06, 2005 at 10:42:11AM +0100, David Masure wrote:
Second Post !!! Please help !
Hi,
I'm using Asterisk with a BRI Card (HFC Chipset) using the zaphfc
driver.
What versions of asterisk, bristuff, kernel and linux?
I'm encountering the
You built zaptel vs. the wrong kernel headers?
Could you please provide more details on your build system?
I use ALT Linux 2.4. It's rpm-based distributions with apt. I have
installed this build tools:
# gcc -v
Reading specs from /usr/lib/gcc-lib/i586-alt-linux/2.96/specs
gcc version 2.96
Here are the versions :
Asterisk 1.0.6
Bristuff 0.2.0-RC7k
Kernel 2.4.20-8 on RedHat 9
I must also tell that I have 8 identical configurations running and I
have only one computer doing this problem...
I'm facing the problem when at least one of the two lines is already on
call
Best regards
I
have aproblem with the cdr.
We terminate through
a pstn provider to the pstn network.
The problem is now
the cdr accounts the connection to the gateway. Coz the gateway is answering our call and then forward to the pstn
number.
So i have billsecs
all the time even it is only ringing or so.
Hi all,
I currently see some strange behavior when trying to call an external (NAT)
SIP extension when connected with an IAX phone on a local segment.
When dialing the SIP extension , it only works the first time (or sometimes
2 times) and after that , I only get one-way audio.
IAX2 to IAX2
Hi,
I really like DIAX and i was to stick to it so if you can help solve
my
problem with diax???
I'm not sure that the problem is DIAX related
You are the first one with this issue...
Pls try to use the examples in DIAX help file for iax.conf and
extensions.conf
Best regards,
Dan
Hi friends,
Still i did not receive any instruction about my problem. I reached
somewhere but still my Asterisk not start to work as H.323 Gatekeeper.
I used the following configuration and i found that OH323 is
registered when asterisk starts.
here is my oh323.conf file's configuration
Hi list.
== Primary D-Channel on span 1 up
that repeat all the time in the CLI.
i've got a TE110P working as E1
ztcfg -vv show me this
[EMAIL PROTECTED] ~]# ztcfg -vv
Zaptel Configuration
==
SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
Channel map:
Hello,
I have set up an Asterisk server, and connected it to a Brooktrout
faxcard via PRI. I have 8 extensions configured, 4001-4008, and 4001
goes to channel 1 etc. When I send a fax to 4001, I want it routed with
the number dialed (4001), but all Asterisk sends is the channel number
(1). This
It is also woth checking that your X-Lite setups have Silence
supression disabled. This is enabled by default I believe, and does
cause Asterisk problems.
Regards,
Steve
On 12/5/05, Chuck Bunn [EMAIL PROTECTED] wrote:
Hi,
Two things does your codec set in X-lite match what is set in the sip
On Dec 5, 2005, at 9:02 PM, James B. MacLean wrote:
Paul Redstone wrote:
Hi
We're using three line SIP phones (X-lite), very nice, with
Asterisk 1.2
But we want to prevent either direct incoming calls or calls from
other extensions from ringing if the user is
in another incoming call
Date: Mon, 5 Dec 2005 21:48:56 -0800 (PST)
From: ha i [EMAIL PROTECTED]
Subject: [Asterisk-Users] EAGI Audio Capture
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
Hello Everyone,
Why EAGI is made so complex? The audio captured with
You have the contact set to the extension, you need the contact set to
whatever you dial to retrieve your voicemail. i.e. the one that runs
voicemailmain.
Brent Bloodworth wrote:
Actually I think that is how it is setup now. I configured the phone
through the web interface. Callback mode is
Hi all,
Im testing canreinvite = yes in my sip.conf
with snom190 and a Atcom320.Atcom320 seems support re-invite, but the snom190?
Does anyone known if this phone support it?
How I can be sure that it works?
Giordano
___
funny guy wrote:
Just wondering, is the echo canceller in the TE411P capable of
cancelling the echo caused by the delay over satellite link (i.e. approx
400 ms delay)?
Does anyone have any success story to share?
I'm kinda stuck with a really2 annoying echo... adjusting the gain
Iam using Fedora 3 and gcc is installed Please let me knowMark Quitoriano [EMAIL PROTECTED] a écrit: Jourdan,What Distro are you using? do you have gcc installed? On 12/6/05, jourdan lemieux [EMAIL PROTECTED] wrote: Any help on this please Hi, I am getting this error when comp
On Tuesday 06 December 2005 00:58, funny guy wrote:
Just wondering, is the echo canceller in the TE411P capable of cancelling
the echo caused by the delay over satellite link (i.e. approx 400 ms
delay)?
No. Use MARK2, KB1 or MG2 and enable the agressive mode -- this converts your
link to a
I want to allow my users to be able to
Call Forward Unconditional
Call Forward Busy
Call Forward No Answer
And curently I am doing this via my ATA and phone settings, however
this has the problem that when a call is forwarded it goes out without
an accountcode (Even though the ATA is forwarding
lokotes wrote:
Thanks, your are absolutely right - I was thinking of REGISTER.
I couldn't find any information about that - if it's a known problem why
it's so hard to find any info?
Because we don't spend hours every day writing pages and pages of lists
of things that we know need to be
Hi,
Please excuse the cross post but these seems to be one of those issues
that may be answered by a developer or someone with direct
administrative knowledge of the deep workings of Asterisk. I have
deleted my log files expecting them to be recreated by Asterisk 1.2 but
nothing happens
I'm trying to use Dial with the g option. If i understand the docs
correctly, when the dialed number hangs up, control returns to the
initial context. In other words, the original caller shouldn't hangup,
just the destination and context execution should resume with the next
level.
However, in
Posted a couple of weeks ago. Would be most grateful if some kind soul shed
light on this please?
Hello all,
Since upgrading a couple of our servers to 1.2 I've noticed problems when
talking to users on 1.0.9 servers. The servers are connected via IAX2 with
trunking and jitter buffer enabled
In your shell script that converts the file:
#!/bin/sh
FAXFILE=$1
EMAILADDRESS=$2
CALLERID=$3
tiff2ps -2eaz -w 8.5 -h 11 $FAXFILE | ps2pdf - $FAXFILE.pdf
mime-construct --to $EMAILADDRESS --subject Fax from $CALLERID
--attachment $CALLERID.pdf --type application/pdf --file $FAXFILE.pdf
rm
On Tue, 2005-12-06 at 15:14 +, Chris Bagnall wrote:
[snip]
Since upgrading a couple of our servers to 1.2 I've noticed problems when
talking to users on 1.0.9 servers. The servers are connected via IAX2 with
trunking and jitter buffer enabled (jitter buffer on default settings).
Afaik best
Doug,
When you use Realtime SIP Peers, registration information is stored in the
mysql database, the astdb database, as well as being cached in memory. When
a SIP lookup is done, and registration information is stored in cache, that
information will be used rather than querying sip_buddies. We
Thanks for the reply Anish.
I turned off the caching with rtcachefriends=no, verified the registrations
where not stored in astdb, and saw select queries being performed to the
database, and it still failed. Did you ever try that?
Douglas.
-Original Message-
From: Anish Basu
Jerry Geis wrote:
I have a handful of phones that work with 1.0.9. I was trying to upgrade
to 1.2
and the UIP200 phones dont ring.
below is my config for 1 phone.
I tried it with and without the qualify=yes or qualify=no and did not
seem to make
a difference. still no ring.
Any ideas on
Actually this is completely screwy. I have rtcachefriends=no and Asterisk is
still populating the astdb file. Why the hell is it doing this?
mysql select * from ast_config;
++++---+--+--++-+
| id | cat_metric |
List:
Does anyone know if the new Digium 2400 series cards will
work with Credit Card Processing Terminals (commonly found at the checkout). If
these cards do not work, is anyone currently employing another solution?
Thanks,
Garrett Smith
[EMAIL PROTECTED]
Ask Me About Our
This is an outbound issue that affects SIP and Zap (T1 from another PBX)
channels going out our PRI to Telco.
I have two ATT conference number that will take the conference access
codes. (in theory)
(214) 622 4991
(866) 340 2763
If we dial the toll free one, the menus time out because they are
Hello
I Have a machine (P3) acting like a E1 - SIP gateway
(with a digium TE110P)
On this asterisk we are running an AGI doing radius
acounting (it works very well!)
But now we need to make effort test of the hardware
we use.
How we can simulate many concurrent calls? Has anybody
has some clue.
Well... not so perfectly.
What I'm experiencing is that during certain call volumes, many calls
go thru from box1 to box2. However, there are some cases where I get
this message:
Dec 6 11:11:19 WARNING[203]: chan_sip.c:9525 handle_response_invite:
Forbidden - wrong password on
I mistakenly followed a how to guide on voip-info.org describing how to setup the 501s with [EMAIL PROTECTED] It appears that you have set me on the right track as setting the contact set to *98 brings up the voicemailmain. The next logical question is - How do I setup the contact to enter the
Hey everyone,
I'm having a slight problem with my dialplan which
I was hoping you could help me with.
First let me explain the scenario;
I have a few hundred different customers split
into a few different area codes. What I want to
allow them to do is to call each other normally,
ie with area
Don't want to point out the obvious, but seems to me that the lowest common
denominator here is to dial out the PRI if there's no extension match,
correct? If this is the case, then you can use the 's' extension. The 's'
extension is a 'match-none' extension and is invoked when there is not match
i have a software in visual basic anybody know a activex or dll to handle
the asterisk???
thsnk in advance
Vladimir
__
Visita http://www.tutopia.com y comienza a navegar más rápido en Internet.
Tutopia es Internet para todos.
Asterisk.Net, there is a page on the Wiki.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vladimir
Montealegre
Sent: Tuesday, December 06, 2005 8:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] activex
i have
Title: Message
Hi,
What
application can I use to stream the audio for "streaming audio
MOH"?
Regards,Stojan
Sljivic
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stojan
Sljivic - GDSSent: Monday, December 05, 2005 14:24To:
thanks kerry but you have the link?
- Original Message -
From: Kerry Garrison [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Tuesday, December 06, 2005 11:57 AM
Subject: RE: [Asterisk-Users] activex
On Mon, Dec 05, 2005 at 02:24:00PM +0100, Stojan Sljivic - GDS wrote:
Hi,
Have someone successfully configured the streaming MOH in Asterisk 1.2.0
using streamplayer?
streamplayer is basically netcat with many of the options removed,
right? nothing special about it. Any real reason for
Vladimir Montealegre wrote:
thanks kerry but you have the link?
http://www.voip-info.org/wiki/view/Asterisk+.NET
--
Best regards,
Bartosz Piec
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Hi !
I'm planning to replacea legacyPanasonicPBX with Asterisk, but there are 2 issues to resolve before that.
For default, the extensions only can dial to local numbers, but when they want to call to cell phones, long or international phones, there are authorized users, each one with their own
thanks for the reply
- Original Message -
From: Bartosz Piec [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 06, 2005 12:11 PM
Subject: Re: [Asterisk-Users] activex
Vladimir Montealegre wrote:
I have some phones that perform better with rfc2833 for
DTMF, but a termination provider that only supports INBAND.
Is this possible;
Phone G.711u/SIP/RFC2833 DTMF Asterisk
G.711u/SIP/INBAND DTMF provider
If so what are the relevant things to check, right now it
fails, assuming it
One of the benefits of that method is that in your dialplan, you can say
is the caller id of the originating call the same as the extension they
tried to dial ie. I'm number 112 and I dial 112 from my phone. the
dialplan realizes this, and instead of sending me to my voicemail, it
sends me
I don't know where you're based, so I've no idea if this'll work for your
users.
If you're using GSM mobiles there are a load of (reasonably) standard
vertical service codes to enable/disable call forwarding etc. depending on
conditions. How about this:
1) set up a DID that's never answered in
is any problem with faxing trought:
PSTN FAX = PRI = ASTERISK = SIP/G711 = SIP ADAPTER
(like Linksys PAP2 etc.)
--
[EMAIL PROTECTED]
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You can accomplish password per extention by using an AGI script.
model would be,
keep extension and password in a table
Execute a simple script to authenticate before dial-out
You can also accomplish dial-out time from an AGI script.
Feel free to ask if you need further help.
Thanks,
--You
Hi all,
Got the snom 320s, and I love them. Only issue I'm having with setup is
getting the retrieve button working. I have specified in my sip.conf:
[EMAIL PROTECTED]
Which is my Checking voicemail extension. However, when I hit the
'Retrieve' button, it seems sporadic what it dials.
Hi,
For default, the extensions only can dial to local numbers, but when
they want to call to cell phones, long or international phones, there
are authorized users, each one with their own password for dialing.
I've checked the password for outgoing routing in Asterisk, but the
password
On 10:03, Tue 06 Dec 05, Sean Kennedy wrote:
Hi all,
Got the snom 320s, and I love them. Only issue I'm having with setup is
getting the retrieve button working. I have specified in my sip.conf:
[EMAIL PROTECTED]
Which is my Checking voicemail extension. However, when I hit the
Several times a day I get this happening when I try to dial out. Is
there something on your side limiting concurrent calls or is it in my
config
Thanks for any help,
*
Called voipjet_out/XX
-- Call accepted by 64.34.45.100
Using AGI is overkill as everything can be accomplished in the DP.
Use the VMAuthenticate command for the passwords:
http://www.voip-info.org/wiki-asterisk+cmd+vmauthenticate
and use include based on time for the other stuff:
http://www.voip-info.org/wiki-Asterisk+tips+openhours
On 12/6/05,
Yeah, it shoud NOT work 100% of the time (maybe not even 50%)
On 12/6/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
is any problem with faxing trought:
PSTN FAX = PRI = ASTERISK = SIP/G711 = SIP ADAPTER
(like Linksys PAP2 etc.)
--
[EMAIL PROTECTED]
Hi Steef!
Does anyone have some ideas on how to setup a shared line on several
SNOM phones in a reliable manner?
The description of what exactly you are trying to accomplish is a bit
scarce, which makes good suggestions a bit difficult... ;-)
Calls enter on number 123. They do not have to
Garret, if it uses the same technology that the T1 cards from digium
use (which I assume it does, it's bridged internaly), then it should
work without a problem. In any case I like better a Digium single span
T1 card with an Adit 600. The only advantage of the 2400 is that it
allows you to do per
I just hard set it in the SNOM embedded webpage. My VM exten is *98 and I
set it to that in the SNOM webpage and it works 100% of the time. Might be a
bit of an unmanageable problem, though if you have a few hundred extensions.
-Original Message-
From: Sean Kennedy [mailto:[EMAIL
Create a context for that ATA that always applies the account code in
the DP before it you issue the dial command.
On 12/6/05, Matt [EMAIL PROTECTED] wrote:
I want to allow my users to be able to
Call Forward Unconditional
Call Forward Busy
Call Forward No Answer
And curently I am doing
I don't know about your echo problem, but when you have such high
latency please don't call me :)
On 12/6/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 06 December 2005 00:58, funny guy wrote:
Just wondering, is the echo canceller in the TE411P capable of cancelling
the echo
I use SetAccount(${EXTEN}) when the extension gets the call. The original dialed extension will be recorded as AccountCode in CDR, before the call is forwarded. The 1st field in CDR will be the extension your customer, the 2nd will be the caller (source), the 3rd will be the forwared number.
It
Hi!
Does anyone have some ideas on how to setup a shared line on several
SNOM phones in a reliable manner?
Here's an option I forgot:
Put the incoming caller into a MeetMe room and then ring whatever
internal phones you'd like to ring. Use app_devstate to play with the
lights as
Tony Hoyle wrote:
btw. does anyone have a definitive list of all the finarea VOIP
companies? I can think of:
call1899
call18866
voipbuster
sipdiscount
voipcheap (note: this one uses a proprietary protocol, similar to IAX
but over different ports and not compatibile).
Hi !
I was reviewing the tips+openhours and it's clear for me. But i don't figure out how to use cmd+vmauthenticate for extension password for outgoing dialling.
Any help would be appreciate
Thanks.
2005/12/6, C F [EMAIL PROTECTED]:
Using AGI is overkill as everything can be accomplished in the
I have a toll-free number that is mapped to the main number of my PRI.
When a call arrives, the called number is the main number, not the
toll-free number.
The PRI vendor is ICG, and they're saying the number gets mapped to the
main number. I'm saying I want to see the toll-free number. Can
I recently signed up with TelIAX for voip service.
Im currently using IAX to connect to them. Would connecting to
them via SIP provide less latency or be better in any way? Thanks for
everyones 2 cents!! I just to be sure Im using the
best/fastest trunk methods.
-Ross
Hello Michael,
It was a big pain for my provider as well. I ended up having to burn a
local DID for each toll-free DID to differentiate.
You can ask them to display the toll free number in RDNIS. My provider
wanted stupid dollars to do it.
--
Richard Cook
[EMAIL PROTECTED]
T: 705-223-2000
Wayne Gemmell writes:
Forgive me if this is old news...
http://www.spectrum.ieee.org.nyud.net:8090/oct05/1846
which links to an article about Narus, Inc.'s VoIP blocking
software, and other measures ISPs take to interfere with customer's
VoIP use.
I think that at least in democratic Western
Kind of depends on what you want to do!
Remember Asterisk is not a SIP proxy so if you want to be able to call a
phone from another SIP phone out in the world you probably best off with
ser as a sip proxy and the asterisk as gateways, features servers.
We do a lot of the routing and so with
Hi,
I'm using asterisk-1.0.9 with zaptel-1.0.9.2 and libpri-1.0.9
I have observed that in case of an incoming call to the Asterisk the Q931
message CONNECT (sent from Asterisk after the call has been answered) does
not contain the Connected number information element. This is bad if the
caller
On Tue, Dec 06, 2005 at 09:47:49AM -0700, Colin Anderson wrote:
Don't want to point out the obvious, but seems to me that the lowest common
denominator here is to dial out the PRI if there's no extension match,
correct? If this is the case, then you can use the 's' extension. The 's'
extension
On 12/6/2005, Code Lover [EMAIL PROTECTED] wrote:
Hi friends,
Still i did not receive any instruction about my problem. I reached
somewhere but still my Asterisk not start to work as H.323 Gatekeeper.
I used the following configuration and i found that OH323 is
registered when asterisk starts.
hi to all
again
how i do to receive fax and the faxes and the incoming fax put in directory
/xxx or any directory in tiff format???
- Original Message -
From: Wolf N. Paul [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 06, 2005 2:35 PM
Subject:
Hrmm that works except that my accountcode is not the extension of the
customer/user, but is a distinct accountcode (ID).
Oooo... you are setting the accountcode when you GET the
call. I guess I could do that... before I go to do too much work, is
there a way to get asterisk to know
Add this to your phone1.cfg:
msg msg.bypassInstantMessage="1"
mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact"
msg.mwi.1.callBack="*98805"
msg.mwi.2.subscribe="" msg.mwi.2.callBackMode="disabled"
msg.mwi.2.callBack=""
msg.mwi.3.subscribe="" msg.mwi.3.callBackMode="disabled"
I'm not sure what you are trying to set it to, I'm assuming that some
of the stuff you want is available here:
http://www.voip-info.org/wiki-asterisk+variables
or in README.variables in /usr/src/asteriks (or one of the sub folders)
Look at RDNIS or DNID, either one might have the dialded number
I have customer wtih 30 stations in cubicles but they only
have 1 rj45 per cubicle and that is for lan and internet.
I would prefer the voip to be on separate net connection for quality
purposes
but customer does not want to recable. How to avoid voice quality problems
?
I have read about
This can be done. Just make sure you specify the DTMF mode you are using per SIP account. for example:
[sipuser]
type=friend
secret=blahblah
qualify=yes
nat=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
etc.
etc.
[sipprovider]
type=friend
secret=blahblah
qualify=yes
nat=yes
disallow=all
As long as it is a decent 100mbit switched LAN and your switches are not
saturated with traffic, you should just be able to plug in the phones and it
will work fine, no prioritization is necessary. A 100mbit full duplex
connection has a potential bandwidth of a few orders of magnitude above what
a
Hi,
Any pointers on implementing outgoing fax detection?
Thanks,
Andre
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Hi Guys,
Can you guys please provide me with some example code
for EAGI to give out GOOD audio data? The audio
returned by perl-script (RAW, gsm and wav) is really
bad and is not clear at all.
Looks like the data returned is signed SLINEAR 16-bit
data. How can i get the clear noise free audio
Hello A_Navone,
On 06-Dec-2005 21:11, A_ Navone wrote:
I have customer wtih 30 stations in cubicles but they only
have 1 rj45 per cubicle and that is for lan and internet.
I would prefer the voip to be on separate net connection for quality
purposes
Well I can imagine, or even to protect the
Title: Re: [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?
check out www.exinda.com if you are looking for a cheaper
solution to Packeteer, also offers more functionality as the design is third
generation.
Cheers,
Dean
From: [EMAIL PROTECTED] on
behalf of Stijn
Don't use Packeteer, we have used that in the past (although not for voip
application) we had too many complaints from our subscribers. We are an
ISP.
-Original Message-
From: Stijn Jonker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 06, 2005 4:15 PM
To: Asterisk Users Mailing List
What version of Asterisk are you running?
We found that some of the older versions of Asterisk did not obey the wrapup
time correctly, especially when the agent was in more than one queue.
PaulH
- Original Message -
From: Adrian Carter [EMAIL PROTECTED]
To:
I never could get voipjet to work (same error) and they wouldn't respond to my email (going on four days now). I got tired of waiting and noticed many others on various forums complaining about their lack of response so I tried Teliax. I got an IAX2 trunk going in less than half an hour start to
Right, but I can't create a context for every device :)
On 12/6/05, C F [EMAIL PROTECTED] wrote:
I'm not sure what you are trying to set it to, I'm assuming that some
of the stuff you want is available here:
http://www.voip-info.org/wiki-asterisk+variables
or in README.variables in
I had an issue with them with passing my callerid improperly and the
call would not go through like you are mentioning.
On 12/6/05, David K Parker [EMAIL PROTECTED] wrote:
I never could get voipjet to work (same error) and they wouldn't respond to
my email (going on four days now). I got tired
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