Hi list,
I have an asterisk server with a Zap 4 port FXO card connected to the PSTN,
all of the clients are SIP softphones (I have tried LIPZ4 and kphone). I
have successfully configured asterisk to route incoming calls from the PSTN
to an extention that is a SIP softphone and the user can answer
Hi- let me start off by saying I'm a newbie to Asterisk and this list
and I'll also apologize up front for stupid questions.
I have Asterisk running and 2 SIP phones (X-Lite) plus an iaxtel
gateway set up. I used the configurations from the O'Reilly article and
I haven't even set up voice mail
Thanks for the reply. All of the SIP phones and the Asterisk server are
on the local network (192.168.1.x) on the same side of the router (and
yes the router does have a NAT firewall). I would think the XLite to
XLite would work but it doesn't (yet). I am seeing errors on the
console when the
Hey All,
We are using Asterisks as a voicemail only application, and
so far all is great. (Excellent product!)
However, I do have one question that I am hoping you might
be able to help me with.
In our asterisk application. When our customers call
*55 (our dialplan code to check
This is a fairly simple thing to do. You don;t say what type of phones you are using,
so I;ll assume SIP for the example:
Step 1:
Put
callerid=Darren 1234
for each phone definition in sip.conf, obviously replacing Darren with the user eg
Darren Nay or Joe Bloggs, then replace the 1234 with
On Apr 7, 2004, at 4:23 PM, Darren Nay wrote:
My question is. Is there a way to make asterisk aware of the
calling-from (callerID) number so that it will automatically detect
the number and then go directly to asking them to input their
password.
From show application VoicemailMain try:
Hi all;
I am planning a PBX/Voice mail system for a small business (approx 12
employees with phones). They have an inbound ISDN PRI, which is
probably irrelevant because all inbound calls are routed first to
receptionists which rarely route the calls on (client is a medical clinic).
Any idea
Or, you could apply my patch, that I've been upgrading on the asterisk bug
site.
Check http://bugs.digium.com/bug_view_page.php?bug_id=752
Steve (and anybody else who may know about this code),
I have the code for your privacy enhancements compiled and installed. My
one question is, how do
On Wed, 2004-03-10 at 20:30, Steven M. Sokol wrote:
Or, you could apply my patch, that I've been upgrading on the asterisk bug
site.
Check http://bugs.digium.com/bug_view_page.php?bug_id=752
Steve (and anybody else who may know about this code),
I have the code for your privacy
Hi,
Please excuse me if my question seems too simplistic. I have been reading
the mailing list for some time and I am still a bit confused. Here is the
scenario that I would need to achieve and am wondering if asterisk is the
correct software to use.
(h323) (h323/SIP)
Hi,
Please excuse me if my question seems too simplistic. I have been reading
the mailing list for some time and I am still a bit confused. Here is the
scenario that I would need to achieve and am wondering if asterisk is the
correct software to use.
(h323) (h323/SIP)
--On Saturday, January 10, 2004 11:45 AM -0700 Steve Murphy
[EMAIL PROTECTED] wrote:
Hi!
Does * have the capability to screen calls? IOW, if someone calls in
from outside (ie. not a local extension), can * ask the calling party
to state their name, record it, ring the recipient, play the
--On Friday, January 09, 2004 10:11 PM -0600 Alan Andrews
[EMAIL PROTECTED] wrote:
On Fri, 2004-01-09 at 20:55, Ken Alker wrote:
Does * have the capability to screen calls? IOW, if someone calls in
from outside (ie. not a local extension), can * ask the calling party
to state their name,
Does * have the capability to screen calls? IOW, if someone calls in from
outside (ie. not a local extension), can * ask the calling party to state
their name, record it, ring the recipient, play the caller's name for the
recipient, then give the recipient the choice of answering or forcing
I've been looking at Asterisk for a replacement for our phone system and I'm
hoping someone can help validate my assumptions.
We have 4 analog lines coming into the building. These lines are simple POT
lines and we have them in a hunt group with Verizon so that when a single
phone number is
M. Matt Colgin wrote:
I've been looking at Asterisk for a replacement for our phone system and I'm
hoping someone can help validate my assumptions.
I'll try.. :)
We have 4 analog lines coming into the building. These lines are simple POT
lines and we have them in a hunt group with Verizon so
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Thursday, 8 January 2004 8:32 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie Question-Looking for Feedback
M. Matt Colgin wrote:
I've been looking at Asterisk for a replacement for our phone system and I'm
hoping
Back in December there was a thread or remark about a Digium quad FXO
card. I would like to know when Digium will start marketing this...
M. Matt Colgin wrote:
I've been looking at Asterisk for a replacement for our phone system and I'm
hoping someone can help validate my assumptions.
We have
-Pro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Wednesday, January 07, 2004 1:32 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie Question-Looking for Feedback
M. Matt Colgin wrote:
I've been looking
Hi!
First, I want to let everyone that I am new to the Asterik world and
looking forward to have fun with it.
I am interested in setting up a small pbx. The following are some
pre-requisites:
i.) Would like to have three extension with the capability of expanding
using Hard VoIP phones.
Is there any one out there using an MSI 240 Global Station with Asterisk? I didn't
see it listed on the hardware page but figured I would ask just in case.
Thanks,
Patrick
winmail.dat
On Thu, 2003-10-16 at 16:29, rnc Info Lists wrote:
Seems you used my abreviation. It is really known by zaptelrtc. It seems
to be written by Klaus-Peter Junghanns [EMAIL PROTECTED] and is
distributed at http://www.junghanns.net/asterisk/.
Thanks for the info Steve. I got it but the
look at the rtc driver then. you do have a rtc chip already on the
system.
I looked back in the list and looks like the message that mentioned who
wrote ztrtc I deleted. Can someone please let me know where to obtain
ztrtc? I did a google on it and came up empty.
Thanks,
Robert
http://www.junghanns.net/asterisk/downloads/zaprtc.0.0.1.tar.gz
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of rnc Info
Lists
Sent: Thursday, October 16, 2003 19:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] newbie question: Meetme (looking
Seems you used my abreviation. It is really known by zaptelrtc. It seems
to be written by Klaus-Peter Junghanns [EMAIL PROTECTED] and is
distributed at http://www.junghanns.net/asterisk/.
Thanks for the info Steve. I got it but the make didn't work. Will work
on it over the weekend.
Not
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference room the following message is played:
That is not a valid
On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference
On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the quality of the audio.
When I dial into the conference
Did you modprobe ztdummy before running asterisk ? I have meetme
running in one * box without zaptel harware.
rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to
On Wed, 2003-10-15 at 14:48, rnc Info Lists wrote:
On Wed, 2003-10-15 at 14:16, rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to see how it works not the
Did you modprobe ztdummy before running asterisk ? I have meetme
running in one * box without zaptel harware.
I just tried that.
The following messages are given:
/lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No such
device
Hint: insmod errors can be caused by
On Wed, 2003-10-15 at 15:10, rnc Info Lists wrote:
Did you modprobe ztdummy before running asterisk ? I have meetme
running in one * box without zaptel harware.
I just tried that.
The following messages are given:
/lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: init_module: No
On Wed, 2003-10-15 at 15:59, rnc Info Lists wrote:
The USB card is the problem.. I should have realized that from today's
other thread... This system has no USB and unless I find a ISA USB card it
won't either since the PCI slots are full.
There is plenty else to get familar with so meetme
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have the following questions:
1.) Can Asterisk interface with the OpenSwitch12 board? I've read some
postings that say yes and some that said no.
2.) If I use a soft phone do I still need to purchase a board like the open
switch or TDM400P to
Softphones like the xten.com's X-Lite are free (as in free
bear) and do not require any hardware on the asterisk server
so thay are a real cost saver but I think you will find your
users will revolt as most people don't like to use a computer
phone and want a real phone on their desk.
One
Chris,
I can't help you with experience with the openswitch12 board.
It sounds like you're wanting a combination of physical analog extensions
and soft phones. Depending on the number of analog extensions you'll
need, look at getting a T400P and use a channel bank or two.
No, you don't need
Hi!
I have an H.323 network. I am trying to do a SIP-H.323 gateway to be able to
accept all the calls that come from other networks (SIP).
I have some questions:
- Must I use IAX? If so, how? I have not SIP servers, I only have a H.323
gatekeeper.
- If not, where is that I say that one call
Just the sort of newbie question we all hate ;-)
I'm a bit stuck with MOH. I think all is done right and I've read
everyhing I can find, but whenever * tries to do MOH, all that happens
is
'-z: No such file or directory'
Yes, I am on redHat. Yes I have installed real mpg123. Yes, it does
]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 01, 2003 5:21 PM
Subject: [Asterisk-Users] newbie question: MOH problem
Just the sort of newbie question we all hate ;-)
I'm a bit stuck with MOH. I think all is done right and I've read
everyhing I can find, but whenever * tries to do MOH, all
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello All,
I have a very newbie question: I recently bought an Asterisk Developer's
Kit (TDM). Which of the following can I use with an TDM400P card and expect
it to work properly:
- - A $5 analog telephone I got from Wal Mart into the TDM400P
On Fri, 29 Aug 2003, Timothy Soos wrote:
-BEGIN PGP SIGNED MESSAGE-
I have a very newbie question: I recently bought an Asterisk Developer's
Kit (TDM). Which of the following can I use with an TDM400P card and expect
it to work properly:
- - A $5 analog telephone I got from Wal
Hello,
I am looking for a pbx solution which is not too expensive
but flexible :-) (a customer is in need of a call center
and crm solution).
The customer favors linux on the call center server and
the crm clients.
On a search for solutions i found Asterisk but it looks as
supports analog phones
Peter,
Did you read the website? Not only does it support h323.
Inter-Asterisk Exchange (IAX)
H.323
Session Initiation Protocol (SIP)
Media Gateway Control Protocol (MGCP)
http://www.asteriskpbx.com/index.php?menu=features
bkw
On Thu, 21 Aug 2003, Peter Eckhardt wrote:
Hello,
I am
Peter Eckhardt [EMAIL PROTECTED] writes:
I just found the draft of the handbook. The software is
amazing
Does anyone use Asterisk in Germany on a BRI (S2M) interface ?
http://www.junghanns.net/asterisk/page1.html
Driver to use * with a capi compliant isdn card. I currently use AVM
On Thu, 2003-08-21 at 17:10, Peter Eckhardt wrote:
I just found the draft of the handbook. The software is
amazing
Does anyone use Asterisk in Germany on a BRI (S2M) interface ?
I'm in the Netherlands, but I use Asterisk on a BRI using a Fritz!Card
ISDN adapter and the chan_capi
Title: RE: [Asterisk-Users] newbie question - devices
Santiago:
Just internally speaking for 20 users with very little room for growth you could purchase a T100P (T1 card) from Digium. Place the T100P it in the Asterisk server. Connect the T100P to a Zhone Z-Plex channel bank (or any other
RE: [Asterisk-Users] newbie question - devicesHi,
So let me understand this better.
Asterisk can use SIP gateways which offer PSTN access. For example
www.iconnecthere.com, can be used?
Is this correct? And if it is, than any incoming calls through that
service, could be redirected
Title: RE: [Asterisk-Users] newbie question - devices
Santiago:
Ok then you can use asterisk as the gateway between the PSTN and an internal VoIP network. I assume you do not want to purchase any analog phones or VoIP phones, just PCs with a good sound card, speakers and a microphone? You
]
Subject: RE: [Asterisk-Users] newbie question - devices
RE: [Asterisk-Users] newbie question - devicesHi,
So let me understand this better.
Asterisk can use SIP gateways which offer PSTN access. For example
www.iconnecthere.com, can be used?
Is this correct? And if it is, than any incoming
On Mon, Aug 04, 2003 at 12:50:34PM -0500, santiago wrote:
MESSAGE with PGP'd part
Please do not post PGP encrypted messages to any mailing list.
A PGP signature depends on your key, not the recipients. Those would be
ok.
--
Scott LambertKC5MLE Unix
Hi!
I've installed Asterisk and connected ATA-186. When I press 8500, I
listen voice main menu and prompt for enter mailbox number. I press 1234,
but asterisk not accept number and switch to demo-instruct.
Also Asterisk write warning:
NOTICE[77839]: File rtp.c, Line 221 (process_rfc3389):
except that I must enter the DTMF codes several
times in order to be correctly interpreted.
The used SIP phone is a Cisco 7960
Dan
- Original Message -
From: Andrey Katkov [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 03, 2003 12:29 PM
Subject: [Asterisk-Users] Newbie
RFC3389 is comfort noise. By default the ATA 186 will generate rfc3389
packets. You can turn this off through the ATA 186 web interface.
It looks as though you need to configure that ATA186 properly - several
people have posted guides on this.
Iain
--On Thursday, July 3, 2003 9:29 am
:53 PM
Subject: Re: [Asterisk-Users] Newbie question
RFC3389 is comfort noise. By default the ATA 186 will generate rfc3389
packets. You can turn this off through the ATA 186 web interface.
It looks as though you need to configure that ATA186 properly - several
people have posted guides
PROTECTED] On Behalf Of Andy Powell
Sent: 06 June 2003 03:33
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie question on soft phones with SIP
and
*
Tielman,
You can take a look at the quick and dirty guide I'm slowly putting
together if you like...
http://www.automated.it
: Re: [Asterisk-Users] Newbie question on soft phones with SIP
and
*
Tielman,
You can take a look at the quick and dirty guide I'm slowly putting
together if you like...
http://www.automated.it/guidetoasterisk.htm
I'd appreciate any feedback you have on it.. and if it helped
Andy
Nice work Andy.
Filled in some issues I didn't quite understand. Looking forward to more.
Chris
Tielman,
You can take a look at the quick and dirty guide I'm slowly putting
together if you like...
http://www.automated.it/guidetoasterisk.htm
I'd appreciate any feedback you have on it.. and
Tielman,
You can take a look at the quick and dirty guide I'm slowly putting together if you
like...
http://www.automated.it/guidetoasterisk.htm
I'd appreciate any feedback you have on it.. and if it helped
Andy
*** REPLY SEPARATOR ***
On 06/06/2003 at 14:17 Tielman
On Fri, 2003-06-06 at 15:32, Andy Powell wrote:
Tielman,
You can take a look at the quick and dirty guide I'm slowly putting together if you
like...
http://www.automated.it/guidetoasterisk.htm
I'd appreciate any feedback you have on it.. and if it helped
Andy
Excellent stuff
On 06/06/2003 at 17:36 Patrick wrote:
Excellent stuff Andy. It was quite a disappointment that the document
stopped before explaining ..errr everything :) Look forward to learn how
to setup one-way conference and music on hold. Thanks for the guide so
far.
Regards,
Patrick
Glad it was of use
I have an ATA-186 in a SIP configuration (following Shawn Djernes
how-to), but I get the following error at the asterisk console when I
try to call the phone connected to the ATA:
ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device
Failed to register zone 'United States / North America':
201 - 261 of 261 matches
Mail list logo
