On Wed, 2003-12-03 at 05:00, Carlos Arnt wrote:
Hi all,
I just trying to test MSN 4.7 that has SIP.
Because with him i can use a video and voice transmission and * .
But when i try to call someone using the DIALPAD of MSN, when i insert
any digit into * the numbers appears twice !!
like
rm -rf reggie
On Tue, 2003-12-02 at 22:35, reginald huey wrote:
Please
Remove me from the list
Reggie
Reginald Huey
__
Do you Yahoo!?
Free Pop-Up Blocker - Get it now
I thought the origin of outbound connections were random, but the
destination was always the port of the service you're attempting to acquire?
That's the case with TCP.
Not UDP.
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hi there,
i've been able to successfully run asterisk with the Voicetronix OpenLine4
card, it can accept calls and function normally. The only problem I'm
experiencing so far is getting the card to outdial to a third party.
What I'm trying to achieve is basically call bridging, where the caller
Hello every one,
I have got a
H323 gatekeeper for testing. The informations are something like
this:
account code: test01
gk ip address:192.168.10.12
I don't know how to set it in the h323.conf or
oh323.conf, I have tried it for almost one day but I always got the error. Help
me please.
All,
Here's a cool one.. I was attempting to call a retarded conferencing
service, and was having problems with it picking up my DTMF.. after trying
all the settings my Sipura SPA2000 offers, I found inband actually works..
unfortunately, I can't get anything else to pick up my inband
Brian West wrote:
If you have echo on the X100P's Mark setup chan_zap to pretrain the echo
can, but it had a few issues until today which Mark nailed down the bug
that caused the DTMF to be unreliable.
Ok here is how you would do it:
Thank you!
Hi All,
I am a newb to *. Just configured and lucklily it
worked on the first attempt. My setup is on Rh 7.2 and i d/led the build on Dec
1st. i hv installed X-Lite on two of my laptops. i am unable to check my
voicemails. when ever i enter my password * prompts me again and again to enter
On Wed, Dec 03, 2003 at 08:30:34AM +0100, Roy Sigurd Karlsbakk wrote:
hi all
I spoke to this guy the other day, working with Cisco's VoIP system. He
told me they were using a PRI/E1 to transport SMS, and could even do so
from their phones.
May this be possible with asterisk? I have an
Hi All,
I am a newb to *. My setup is on Rh 7.2 and i
d/led the build on Dec 1st. i hv installed X-Lite on two of my laptops. i am
able to make calls between X-Lite (Ext 2000 and 2001) . i configured MSN as ext
2002. When ever i am trying to log on using MSN it rejects my password.
the
:) h323.conf is just a bit strange (there is no simple/clear alias
options as in the oh323.conf)
But it's a good idea to read Readme and h323.conf.sample ...
here is one h323.conf
[general]
port = 1720
bindaddr = 0.0.0.0
tos=lowdelay
dtmfmode=rfc2833
context = your-unautorized-context
On Tue, 2003-12-02 at 15:55, Arnold Ligtvoet wrote:
Hi Leif,
I tried the patch. Installed it exactly as described per your email. Thought
that you might be interested that it works for me as well. Like a charm, I
can finally call FWD numbers like 10001 and 612 (speaking clock demo).
BTW:
Prolly change the auth= to plaintext...
On Wed, 2003-12-03 at 10:07, Balaji NJL wrote:
Hi All,
I am a newb to *. My setup is on Rh 7.2 and i d/led the build on Dec
1st. i hv installed X-Lite on two of my laptops. i am able to make
calls between X-Lite (Ext 2000 and 2001) . i configured
Hi,Lubo,
Thank you very much for your reply. I want to use the gatekeeper for
outbound call, but I really don't know how to use it in the extensions.conf
,I think there are something diffrence between the chan_h323 channel and the
chan_oh323 channel. A little example of extensions.conf would
Take a look here, I hope it will help you :)
http://www.voip-info.org/tiki-index.php?page=Asterisk
http://sprackett.com/asterisk/conf/
http://www.loligo.com/asterisk/current/
http://www.fnords.org/~eric/asterisk/
Lubo
[EMAIL PROTECTED] wrote:
Hi,Lubo,
Thank you very much for your reply. I
Hello,
is there a way to make app queue to first try to ring the agents and
start music on hold only when they are all talking to other callers?
So when the caller calls, and there are free operators he hears ringing,
and * is not picking up until call is answere, or specified timeout.
And if
Hello,
After a while the transfer on grandstream stops working, only the reboot
fixes the problem. It also seems that it may be the phone I`m trying to
transfer _to_ also sometimes requires a reboot. After that it starts
working. I`m using RFC2833 signlaing between phones and *. Does anybody
Hi Folks,
I have a X100P with Asterisk running connection to a non-asterisk
device in the other side. It was working perfectly with H323(chan_h323)+
G.729 in the last weeks.
Suddenly, I am getting double 3's in the other side's POTS. Any number
is not repeated, only the 3 is being repeated.
I
Hi,
there is a bug in chan_mgcp.c which shows up if you have more
than one MGCP gateway configured with host=dynamic.
The problem is in the routine find_subchannel when a MGCP
response is received. When the response is handled find_subchannel
is called with name = NULL and sin = address. This
Anton,
Take a look at the latest version of the patch in:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=214
Good luck!
Michiel
Anton Yurchenko wrote:
Hello,
is there a way to make app queue to first try to ring the agents and
start music on hold only when they are all
Hi,
I got this setup.
analog phone (ext7) --- analog pbx - (ext 6 analog) voicemodem (ext 3
asterisk) ttyS0/asterisk sipphones
q1:
I got the voicemodem to work, but oneway only. I can talk from my analog
phone, to my sipphone, but not the other way ? I know it only suppose to
Michiel Betel wrote:
Anton,
Take a look at the latest version of the patch in:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=214
It does adds an abiliti to make an announcment to a user once they are
in queue, but no this behaviour with cheking if all operators are busy
or
How do I get asterisk to populate the Calling Party Number field in an
H.323 call?
I have asterisk configured to accept a SIP call and connect it to an H.323
IVR system. The call goes through, but the caller id is put in the Display
field rather than the Calling Party Number field.
Please help!!!
Anyone have tried * with kerio SIPPS softphone?
It registers ok with *, but
I get missing sdp body message when dialing any extension.
Thanks.
Hector-.
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Hi all,
The new multilingual version of DIAX (0.9.5) is now available for:
- English
- Romanian
- German
- Dutch
- Italian
- French
- Spanish
- Portuguese
at the following locations:
http://www.laser.com/dante
http://www.geocities.com/tdanro
What's new in 0.9.5 :
- double support(IAX(1)/IAX2)
-
Try the following:
vpb.conf:
[interfaces]
echocancel = on
board = 1
context = default
mode = fxo
channel = 3
extensions.conf:
exten = _9.,1,Dial(vpb/1-3/${EXTEN:1})
exten = _9.,2,Congestion
Hope help you
Jorge
Ahmad Faiz wrote:
hi there,
i've been able to successfully run asterisk with the
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 9:04 AM
Subject: [Asterisk-Users] New Multilingual DIAX (0.9.5) available for
download
Hi all,
The new multilingual version of DIAX (0.9.5) is now available for:
-
Roger Workman
General Manager
PCS: 304-751-6286
Fax: 304-399-0046
ICQ: 4447584
This e-mail and attachments, if any, may contain confidential and/or
proprietary information. Please be advised that the unauthorized use or
disclosure of the information is strictly prohibited. If you are not
Bug:
In the Phonebook, (I've only tried it this way, so far) if you delete an
item, then choose a different item from the dropdown, the delete button
doesn't work anymore.
I was deleting the default entries and decided to not delte the DIGI entry.
When I chose the next one after that, it
Hi,
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 4:29 PM
Subject: Re: [Asterisk-Users] New Multilingual DIAX (0.9.5) available for
download
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk
Hi,
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 4:33 PM
Subject: Re: [Asterisk-Users] BUG in New Multilingual DIAX (0.9.5) available
for download
Bug:
In the Phonebook, (I've only tried it this way, so far) if
I don't know if this was intentional or not, but my newest download
defaulted to Romanian?
Sorry for that. It was not intended to be like that.
You can switch to any language you want using CTRL+ first letter of the
language.
For example, use CTRL+e to switch to English.
Then it will be
On Tue, 2 Dec 2003 21:11:31 -0500 (EST), firedude wrote
The new versions of iaxcomm and DIAX are both now using the
iax2 protocol. So in order to receive incoming calls on
either of them in your extensions.conf file change
IAX/clientname to IAX2clientname. Then you should be able
to
I have 40 of these phones. they dont run SIP or any usable protocol they can
hook up to a Nortel box and proxy SIP out of that box, but they wont run SIP
native if im wrong please let me know... I'd relly like to use my 40 phones that
are collecting dust
Dave
[EMAIL PROTECTED]
I repost a message I put a week ago:
I have a Cisco IAD 2431 which has MGCP protocol; I cannot make
it to work againts Asterisk; at least there is some MGCP conversation
between them but when I offhook a phone attached to IAD I get no tone at
all.
As anybody managed to get
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of PJ Welsh
Sent: Tuesday, December 02, 2003 9:39 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] PREPAID APPLECATION
It is a shame that within a couple of hours they can tell you
to remove
Hi,
Maybe someone has seen this before...
I've installed a new T100P, but it doesn't seem to work. I've attached
the T100P to an Adtran 750 using a crossover cable. The Adtran shows a
red alarm on the T1 interface. The Adtran has been set to factory
defaults with FXS cards in 1-3 and an
Hi Mike-
Not sure why your card seems dead, but your second crossover spec seems to
be the correct one.
Here's a link to a good diagram of the crossover cable (see the bottom of
the linked page). Only 1, 2, 4, and 5 are used. It is not necessary to
connect the others.
I know this isn't the Cisco list, but enough people here are wired
into the VoIP world that perhaps someone has heard if Cisco has
released a SIP image for the 7920 yet...
JT
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I am still having these same problems. Anyone with experience with
these apps that could point me in the right direction?
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller
I bet they are going to use SIP at some point.. just not yet.
On Wed, 3 Dec 2003, Gary wrote:
which would make their Multimedia Terminal Adapter an interesting
device ??
On Wed, 3 Dec 2003 10:41:15 +1100, Adam Hart wrote:
did you even read what I said?
but if you look, it's actually
Hi all ( dev user list ),
I'm starting to implement the missing features in Meetme application :
's' -- send user to admin/user menu if '*' is received
Line 438
app_meetme.c
-
else if ((f-frametype
Ok here is a question that has
gotten me stumped. I have an Asterisk system up and running. I need
toconnect it via the Internet to a Sip Cisco system. This is what
they have. I have there IP address's and login. X-lite is able to connect
to them and make a call! So I have the name right!
On 03/12/03 16:43, Steven Sokol wrote:
Thanks, but I already have the clients configured as IAX2 rather than
IAX. The failure is not universal (not ALL calls are missed). Rather
the client seems to go to sleep for some reason -- almost always after
handling a call.
I have been monitoring the
Does Asterisk have the capability to re-route calls that have already been
connected?
By this, I mean:
1. A call comes in to Asterisk.
2. It is routed to an extension as normal.
3. This extension answers, and the conversation starts.
4. After a few minutes, a plugin that I am writing decides
Anton Yurchenko wrote:
Hello,
After a while the transfer on grandstream stops working, only the
reboot fixes the problem. It also seems that it may be the phone I`m
trying to transfer _to_ also sometimes requires a reboot. After that
it starts working. I`m using RFC2833 signlaing between
I have not heard and I was just looking myself. I would say no at this
time, possible 1st QTR 2004
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Wednesday, December 03, 2003 11:26 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Any
I have a 2621 working with asterisk. See below:
sip.conf
==
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
[cisco] ; Cisco 2621 Router
type=friend
canreinvite=no
check the manager interface ... you can transfer the active call to some
other extension. (redirect). If these are zap channels there is
zaptransfer command and zapdialoffhook via the manager.
regards
Martin
On Wed, 3 Dec 2003, Alistair Cunningham wrote:
Does Asterisk have the capability to
Mark Johnston wrote:
Alistair Cunningham [EMAIL PROTECTED] wrote:
I am working on a project on 3rd party call control for a call center, for
which I think Asterisk may be useful. What I would like to do is:
This is something I've given some thought to lately, with the goal of
writing a
Lubomir Christov wrote:
:) h323.conf is just a bit strange (there is no simple/clear alias
options as in the oh323.conf)
But it's a good idea to read Readme and h323.conf.sample ...
; H.323 Alias definitions
;
; Type 'h323' will register aliases to the endpoint
; and Gatekeeper, if there is
Hi!
Does Asterisk have the capability to re-route calls that have already been
connected?
Look at astman and its redirect button, I guess that is more or less
what you want. So: Use the manager interface.
Cheers, Philipp
___
Asterisk-Users
Hi,
- Original Message -
From: Grzegorz Nosek [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 5:08 PM
Subject: Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm,
etc.)
On Tue, 2 Dec 2003 21:11:31 -0500 (EST), firedude wrote
The new versions of
Hi,
I need to know if someone encounters display errors (like the window
displayed partially) when some 'strage' resolutions are used for the display
in Windows XP native theme mode.
Thanks,
Dan
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OK, an answer is in README.variables causes.h...
[7-digit-PRI-Machine-2] ; The machine connected to PRI 2 (on its g1)
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _9NXX,2,gotoif,$[${HANGUPCAUSE} = 2]?9|1
exten = 9,1,Busy
John
original message *
I have
--- Gary Mart [EMAIL PROTECTED] wrote:
Is there a way to make the voicemail message announcement include
the callerid. It would be handy to know who called (well, at least
where the call was from) especially if they just hung up.
I know I can get it from msg.txt but for the lay user it
Greetings,
I'm trying to setup an option in my greetingmenu that would allow the caller to select
this particular option for emergency calls. That option would dial out on an
available PSTN line to a cell phone number.
Currently it is setup as such
exten = 9,1,Dial(Zap/g1/CELLPHONENUMBER
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I would change the option number to something else because 9 is often
picked up in another context as 9NXXNX
You might have to make a sub menu in order to get there, but try using
2-8 for the menu options.
Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227
Lubomir Christov wrote:
[test01]
type=h323
host = 192.168.10.12
context = your-incomming-context
The keyword host in a type=h323 makes absolutely no sense.
Jeremy McNamara
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John Harragin wrote:
OK, an answer is in README.variables causes.h...
[7-digit-PRI-Machine-2] ; The machine connected to PRI 2 (on its g1)
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _9NXX,2,gotoif,$[${HANGUPCAUSE} = 2]?9|1
exten = 9,1,Busy
Added to
On Wednesday 03 December 2003 09:42, Michael Welter wrote:
I've installed a new T100P, but it doesn't seem to work. I've
attached the T100P to an Adtran 750 using a crossover cable. The
Adtran shows a red alarm on the T1 interface. The Adtran has been
set to factory defaults with FXS cards
Hi!
I need help to undestand the options:
externip= static/ dynamic ip? can be a domain?
localnet= internal ip of * machine?
localmask= 255.255.255.0 ?
Thanks
- Original Message -
From: Leif Madsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 7:25 AM
On Wed, Dec 03, 2003 at 05:47:59PM -0200, listas iPfone wrote:
Hi!
I need help to undestand the options:
hello.
externip= static/ dynamic ip? can be a domain?
externip can by an IP address or a domain. it uses gethostbyname(3)
in the code.
localnet= internal ip of * machine?
Can the phone port on the x100p be an addressable extension on asterisk? I
want to plug our conference phone into that phone jack as it is an analog
phone.
Todd Wallace
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Message: 11
From: Juan J. Sierralta P. [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Organization: Telefonica CTC Chile
Date: 03 Dec 2003 12:23:26 -0300
Subject: [Asterisk-Users] Cisco IAD with MGCP
Reply-To: [EMAIL PROTECTED]
snip
hostname 192.168.65.200
[192.168.65.200]
host =
Hi!
for the record:
Put an behind the line?
It does help to get a proper hang up for the client, but there is no
restart initiated at all... looks like now the system calls gets
cancelled due to the fact that the client is gone.
Ah. Then put a 'nohup' in front of it:
that's only a pass-through, no extension (fxs) is provided.
Matteo.
Il mer, 2003-12-03 alle 22:12, Todd Wallace ha scritto:
Can the phone port on the x100p be an addressable extension on asterisk? I
want to plug our conference phone into that phone jack as it is an analog
phone.
Todd
On Tue, 2003-12-02 at 08:27, Richard Alexander wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Tuesday, December 02, 2003 7:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dedicated * voicemail
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Todd Wallace
Sent: Wednesday, December 03, 2003 4:12 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] phone port on the x100p
Can the phone port on the x100p be an addressable
On Wed, 2003-12-03 at 06:34, Hans-Henrik Andresen wrote:
Hi,
I got this setup.
analog phone (ext7) --- analog pbx - (ext 6 analog) voicemodem (ext 3
asterisk) ttyS0/asterisk sipphones
q1:
I got the voicemodem to work, but oneway only. I can talk from my analog
phone, to
What DTMF options are available to me. My carrier is using DTMF relay H245
Alpha
Todd Wallace
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I'm trying to put multiple Linphones and Snom 200's into a Meetme room.
With two devices, echo is quite noticeable. With 3 or more it
degenerates into white noise. Which part of the software is responsible
for echo cancellation in a MeetMe room? Is it a setting on the phones
themselves, or
On Wed, 3 Dec 2003 20:45:21 +0200, Dan wrote
Hi,
- Original Message -
From: Grzegorz Nosek [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 5:08 PM
Subject: Re: [Asterisk-Users] Iax Client Library Issues?
(DIAX, iaxComm, etc.)
On Tue, 2 Dec 2003
I got an email from him this morning, and I quote:
Hi Aaron,
We are expecting a large container of GS product at the end of this week or
Monday next week. This will clear all backorders that are currently in the
system.
BT-101, BT-102 and HT-286 products are in this container.
Thank you for
Leif wrote:
Awesome! Have you tried the newer patch / diff for 1.259 (which as of
right now is the newest chan_sip file). If you goto bugs.digium.com and
login anonymously and jump to bug 104, then you can get the newest
patch. Same instructions as before.
Installed the new patch, no
Good to hear...
bkw
PS when you think about Asterisk do you touch yourself? :P
On Thu, 4 Dec 2003, Aaron Martin wrote:
I got an email from him this morning, and I quote:
Hi Aaron,
We are expecting a large container of GS product at the end of this week or
Monday next week. This will clear
Anyone wishing to help build/manage openenum.net please contact me via
email [EMAIL PROTECTED] ... I would like to have someone assist in building
and management.
Thanks,
bkw
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Hi
I have a second line that we use for a fax server
Since we are luck to get 2 faxes a week
I want to use this line as a dial out line for *
But still need to be able to send and receive faxes on it
Has anyone got any ideas how I could accomplish this ??
Regards Mick
Okay...here's one for all of you
3 party meet-me conference:
Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM,
no VoIP at all involved. No echo at all.
Call 2: Call comes in via IAX(TDM - Asterisk_1 - IAX/GSM -
MyAsterisk. Caller immediately hears his own echo
Has anyone used the speakers on sip phones as part of an intercom?
Are there sip messages you can send a phone to simulate key strokes,
like someone hitting the speaker phone button on a GS?
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
Thank's Lubomir and Jeremy! It's working now. That's to say,I could dial
long distance call from MSN or NetMeeting now.
Regards.
frank
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003
Oops, my bad.
Turns out it was just mixer settings, feeding back through the soundcard. Sorry for the noise.
Message: 14
Date: Wed, 03 Dec 2003 17:43:16 -0500
From: Matt Lawson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Echo cancel in MeetMe?
Reply-To: [EMAIL
On Thu, Dec 04, 2003 at 10:50:57AM +1030, [EMAIL PROTECTED] wrote:
Hi
I have a second line that we use for a fax server
Since we are luck to get 2 faxes a week
I want to use this line as a dial out line for *
But still need to be able to send and receive faxes on it
Has anyone got
I have used this device with good results:
http://faxswitch.com/stick_fax_phone_modem.html
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Thanks for that
One question how do I stop * from picking up that line
But still allow it to dial
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Wood
Sent: Thursday, 4 December 2003 11:25 AM
To: [EMAIL PROTECTED]
Subject:
Silly question: what kind of phone was the person in California calling on?
Some phones give a local echo while you talk. If that happens, then I
could see it causing problems...
Just a thought... I hope it helps!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
My phone number is being hosted by a provider and brought inbound on a Cisco
5300. A Nextone softswitch is in the middle passing the inbound call to me
as a SIP request to my * box. He shows he is sending me the DTMF's, but I
am not picking them up and interpreting them. I have tried info,
Hello,
I was demonstrating Asterisk capabilities with a SIP Soft phone to
a Key system installer yesterday, and we were discussing where Asterisk
can fit into that market. He brought up some interesting, user-centric
questions which I couldn't answer. I didn't find anything in google
Hi,
I have the VIA chipset, and I'm trying to disable the sound and enable a
soundblaster compatible card.
Can you tell me what you did in /etc/modules.conf to enable your
soundblaster card?
Thanks,
Mike
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[EMAIL
Several things conspired to muck things up the last 3-4 weeks.
1. Surgery (repair of a previous hernia)
2. Travel to work at opening our EU warehouse
3. TSA dropping my laptop, thus breaking my access to our VPN
4. New PRI going to a Asterisk box for our PBX and having the PRI be
mucked up.
Looks like your box has been compromised. Try
ls -l `which ps`
You'll probably find an inapropriate date. Whenever I've diagnosed
problems like this, I've found badly installed rootkits.
To address this on my production machines, I'm going to insruct the
router to only allow traffic that is
Not a silly question. I've given that thought.
To be honest, I'm not sure what kind of phone the California or NJ
callers were using. However, we've had numerous conferece calls using
many other services and have never had echo problems.
The problem, in this case, more likely has something to
A good rootkit will also modify the date and time of the replaced binaries
so they will look the same as the original.
Try to replace your ps command with that from a trusted RH9 machine. If
it works ok then you must do a clean install to get rid of the rootkit.
- Original Message -
http://lists.openenum.net
Subscribe to policy if you wish to help with policy and building of
OpenENUM.
Thanks,
Brian
On Wed, 3 Dec 2003, Brian West wrote:
Anyone wishing to help build/manage openenum.net please contact me via
email [EMAIL PROTECTED] ... I would like to have someone assist
Ahh a memory I'd rather forget, unknown to most, John Todd and myself
started a free enum service, similar to what you're doing. (it was called
freenum.org) Unforunately, the project never really got going, due to lack
of time and interest (after thinking it over). I believe it would never have
Asterisk Users,
Does anyone know the URL for the application API for asterisk? I
haven't been able to find any documentation on it.
Thanks,
Jonathan
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
When I place an outbound call via my Cisco Sip devices 7960
and ATA using iconnect or nikotel as my
SIP LD provider, the call connects and then disconnects after a few
seconds. When the call is placed
from an analog extension via the digium tdm40b it
works fine. I have looked at the Debug
Actually an update here.. there is no audio between any of the sip
phones
-Original Message-
From: Kevin
[mailto:[EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 12:02 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Outbound
SIP Call
When I place an outbound call via
At 3:13 PM +1100 12/4/03, Adam Hart wrote:
Ahh a memory I'd rather forget, unknown to most, John Todd and myself
started a free enum service, similar to what you're doing. (it was called
freenum.org) Unforunately, the project never really got going, due to lack
of time and interest (after thinking
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