Andrew Kohlsmith wrote:
CAN I USE/COMPILE ASTERISK
without any telephone/sound card?
I only want to use it as a IP PBX.
This is another topic covered quite often. Do we have this in a FAQ/Wiki
entry yet?
Thank you for the reminder, now it is:
You have to buy a Cisco contract so you can download the files on their site, but here
is a link with the explanations, because once you have the good firmware ... there is
a way to go :
http://www.loligo.com/asterisk/cisco/79xx/
Michael Devenijn
DKMA
Schaarbeeklei 636
1800 Vilvoorde
Tel:
Just turn off the silence suppresion
Van: Jorge Cisneros Flores [mailto:[EMAIL PROTECTED]
Verzonden: vr 28/11/2003 5:57
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] RFC3389 support incomplete
Hi
When
Leif Madsen wrote:
On Thu, 2003-11-27 at 12:03, Mark Spencer wrote:
Yea, cause I used both Kphone and Windows messenger, and they
successfully registered (and subscribed i think) towards asterisk. Using
Kphone I even get a online status on all other users on the asterisk but
no interaction with
Hcqm wrote:
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Hcqm wrote:
CAN I USE/COMPILE ASTERISK
without any telephone/sound card?
I only want to use it as a IP PBX.
Yes, please go ahead.
I'm running Asterisk on both LInux and FreeBSD servers
We are working with realspeak and it is a wonderfull product (even in product) it
supports up to 20 languages and has aquired a really good prod. stability !
Van: Steve Underwood [mailto:[EMAIL PROTECTED]
Verzonden: vr 28/11/2003 4:41
Aan:
Does MGCP transverse NAT? Seeing as the only decent yet cheap IP phone
is the Swissvoice, it would be rather helpful.
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Hello,
is there a way to disable call waiting in sip? I`m using grandstream 101
and even when the phone is in use I hear ringing in the headset. It is
pretty annoying , is there some way to disable this? I cant find
anything like it in the grandstream docs.
Thanks
--
Anton Yurchenko[EMAIL
Michael Devenijn wrote:
We are working with realspeak and it is a wonderfull product (even in product) it
supports up to 20 languages and has aquired a really good prod. stability !
What kind of money we talking for that product?
Thx.
B.
Hi,
I started evaluating asterisk about a week back. I was trying to configure
asterisk as SIP proxy.
This is the setup that I have now.
I have one linux box running asterisk ( say 192.168.68.15 ) and second box
running partysip (say 192.168.68.6).
I am using SJPhone from windows boxes.
I
ranga wrote:
I have one linux box running asterisk ( say 192.168.68.15 ) and second box
running partysip (say 192.168.68.6).
Now this is what I wanted to achieve.
The other sip server ( here partysip) may have many users registered. It
is not possible to make every user's entry into
Hi Sathya,
I bet you use OpenH323 v1.12.0.
Go to v1.12.2 and you will be OK.
There isn't anything wrong with your syntax, it's an
OpenH323 issue.
Michael.
SW wrote:
anyone who can shed some light ? Or oh323 is completely dumped and I should
go to chan_h323 ?
-Original Message-
From: SW
Anton Yurchenko wrote:
Hello,
is there a way to disable call waiting in sip? I`m using grandstream
101 and even when the phone is in use I hear ringing in the headset.
It is pretty annoying , is there some way to disable this? I cant find
anything like it in the grandstream docs.
Thanks
Hi!
I have started development to import the mailinglist archives into a MySQL
database and creating a full text search possibility on this. My questions;
1) Is this already done somewhere else ?
2) Is this of interest ?
I actually just read someone complaining about this today. I'm
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 28, 2003 2:57 PM
Subject: Re: [Asterisk-Users] Asterisk as SIP Proxy
ranga wrote:
I have one linux box running asterisk ( say 192.168.68.15 ) and second
box
running
Hi!
Does MGCP transverse NAT? Seeing as the only decent yet cheap IP phone
is the Swissvoice, it would be rather helpful.
As far as I know: It doesn't since username = IP.
However, there is an RFC enhancement proposed to allow MGCP behind NAT,
status unkown to me.
I did find a note about
Hi!
The other sip server ( here partysip) may have many users registered. It
is not possible to make every user's entry into extensions.conf. Instead,
any mechanism where I can replace 192.168.68.6 with a variable that
represents the 'To' domain will be a great. In simple, I am looking
CAN I USE/COMPILE ASTERISK
without any telephone/sound card?
I only want to use it as a IP PBX.
YES
you can.
how about IAX2 trunking? does this work with ztdummy?
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Message: 1
From: Sergi Gabunia [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Thu, 27 Nov 2003 12:05:15 +0400
Subject: [Asterisk-Users] MGCP problem
Reply-To: [EMAIL PROTECTED]
I have VOIP network built with MGCP endpoints.The manufacturer of =
endpoints is ASKEY. I downloaded latest
Hi John,
On Fri, 28 Nov 2003, Anton Yurchenko wrote:
Hello,
Has anybody been able to get the Mute button work on grandstream? it
simply does nothing. Only Hold is avalable, which is not that good.
Does the GS even HAVE a mute button? The 101's appear not to.
My 100 has MUTE/DEL in
On Fri, 2003-11-28 at 13:24, John Vozza wrote:
On Fri, 28 Nov 2003, Anton Yurchenko wrote:
Hello,
Has anybody been able to get the Mute button work on grandstream? it
simply does nothing. Only Hold is avalable, which is not that good.
Does the GS even HAVE a mute button? The 101's
Hello,
I'm trying to figure out what portions of Asterisk need a lot of CPU
time.
I thought I read somewhere that a Dual P4 2.something will support
approximately 80 calls. Is this based on calls that Asterisk is
actively doing voice processing for (say, Zap channels and voicemail)?
Would a
MGCP uses RTP, like SIP and H.323, and can therefore not traverse NAT
easily
On Fri, 2003-11-28 at 09:18, Andrew Joakimsen wrote:
Does MGCP transverse NAT? Seeing as the only decent yet cheap IP phone
is the Swissvoice, it would be rather helpful.
Hello Everyone..
I just wonder if Asterisk could be interface with the Avaya PBX to its DS1
card partularly the TN2214B? If yes, how should i do it?, just an ordinary
T1 cable, or do I need some other equipment? Also, need to know the pin
configuration for this.
Thanks in advance for the
Hi,
has anyone around here been using/ testing this device? It connects to 1
PSTN and 1 VoIP line, or 2 VoIP lines and allows up to 15 wireless DECT
phones to be connected. Pricing? Experiences?
Greetings, Philipp
http://www.tedas.de/ip_dect.htm
Die TEDAS VoIP DECT PABX ist eine drahtlose
Hi
We interfaced our ASCOTEL PBX to Asterisk. by EuroISDN PRI , DSS1
It works fine on channels 1- 15, but on 17-31 the miststood each other.
Asterisk speaks in Timeslots, the PBX in B-channels
The signalling is ok, but the bridging is shifted. The first incoming connection is bridged to nirwana
On Fri, 28 Nov 2003, Roy Sigurd Karlsbakk wrote:
On Fri, 2003-11-28 at 13:24, John Vozza wrote:
On Fri, 28 Nov 2003, Anton Yurchenko wrote:
Hello,
Has anybody been able to get the Mute button work on grandstream? it
simply does nothing. Only Hold is avalable, which is not that
Hey Daniel,
Try out the latest CVS and let me know if your problem goes away. We
put in a fix for a SIP problem that might be related to this.
Thanks,
--Greg
On Fri, 28 Nov 2003 00:49:20 +0100, Daniel Chabrol wrote:
I was successfully using the BT-100 phone with CVS 11/10. Now that I've
On Fri, 2003-11-28 at 00:42, Carl Youngblood wrote:
Or maybe noise would have to last for more than a certain period of time
before it triggered another waiting sequence. Like, say, if noise lasts
for longer than 2 full seconds or something.
That may be fine. Although you may have
I've been tinkering with ENUM and found that the lack of a debug message in
enum.c that says it has actually succeeded in resolving an address is a bit
of a nuisance. It makes it difficult to see if failures with ENUM are due
to problems with parsing NAPTR records (in enum.c) or mistakes in
Hi All,
I am a newbie to asterisk, and here is my first problem, where I do not
know any further.
I have to grandstream BT100 connected to asterisk. Working fine, for
calling to each other, and to call via a IAX-Link to the outside.
If I try to call the initial demo from the
You can connect the Avaya PBX and Asterisk by T1 as well as you need to do configuration in both side Avaya(Trunk group,dial plan...)and *(dial plan...)
Howard"Joelson S. Apon" [EMAIL PROTECTED] wrote:
Hello Everyone..I just wonder if Asterisk could be interface with the Avaya PBX to its DS1card
On Fri, 2003-11-28 at 06:51, Matthew Asham wrote:
Hello,
I'm trying to figure out what portions of Asterisk need a lot of CPU
time.
Mostly codec translations.
I thought I read somewhere that a Dual P4 2.something will support
approximately 80 calls. Is this based on calls that Asterisk
Carl Youngblood wrote:
What is EAGI? I will probably use festival for the time being, but
I thing that I would eventually like to use ScanSoft's RealSpeak SDK
because it is so life-like. Unfortunately our text alerts are fully
customizeable, so we can't pre-record them.
Beware the
Brian Capouch wrote:
Michael Devenijn wrote:
We are working with realspeak and it is a wonderfull product (even in
product) it supports up to 20 languages and has aquired a really good
prod. stability !
What kind of money we talking for that product?
I think
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roy Sigurd Karlsbakk
Sent: Friday, November 28, 2003 6:50 AM
To: Asterisk Users
Subject: Re: [Asterisk-Users] Re: ASTERISK WITHOUT ANY CARD
CAN I USE/COMPILE ASTERISK
without any
Hi, this is Cesar Rico
I'ma new Asterisk user, I would like toknow how can I develop an
application with voice over IP on H323 protocol, I've read all the
documentation but I've not found the H323 configuration file,could you
send to me a example for this kind offile in order to geta
Thanks Steve, that pretty much answers what I wanted to know.
I was asking more out of general than for a specific deployment, but
if I do have further questions I'll be sure to elaborate. :)
Matthew
Friday, November 28, 2003, 6:41:23 AM, you wrote:
On Fri, 2003-11-28 at 06:51, Matthew Asham
Patrick wrote:
On Fri, 2003-11-28 at 09:14, Anton Yurchenko wrote:
Hello,
is there a way to disable call waiting in sip? I`m using grandstream 101
and even when the phone is in use I hear ringing in the headset. It is
pretty annoying , is there some way to disable this? I cant find
Hi there!
I'm currently considering various PBX solutions for our office telephone
network, and would very much like to use Asterisk. Currently, my
research is incomplete. I have been recommended to use the above cards,
but it is unclear from my Googling whether my configuration will work:
-
On Fri, Nov 28, 2003 at 05:02:57PM +, David M. Wilson wrote:
- 3x 6 b-channels.
- 6 b-channels.
Sorry folks!
David.
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David M. Wilson wrote:
Hi there!
I'm currently considering various PBX solutions for our office telephone
network, and would very much like to use Asterisk. Currently, my
research is incomplete. I have been recommended to use the above cards,
but it is unclear from my Googling whether my
On Fri, 2003-11-28 at 04:15, Cees de Groot wrote:
Leif Madsen [EMAIL PROTECTED] said:
outside_addr=216.239.33.100 ; this can also be a FQDN! ie.
; my.domain.com
Which might be a problem for dynamic environments, but still nice that
someone implemented
Hi All,
Please drop me an email if you can provide Iax termination in India.
PauloHM
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On Fri, Nov 28, 2003 at 06:15:38PM +0100, Peer Oliver schmidt wrote:
As far as I know, AVM only allows a single Fritz!Card PCI in a PC. I
/think/ there is a patch out there to allow more than one. Search the
archives to find out more.
Thanks for the quick response. I'm afraid I was unclear
David M. Wilson wrote:
Hi there!
I'm currently considering various PBX solutions for our office telephone
network, and would very much like to use Asterisk. Currently, my
research is incomplete. I have been recommended to use the above cards,
but it is unclear from my Googling whether my
I have created a script which will install Asterisk from CVS sources
with a single command. This was mainly for my own use so that I could
do either an install or update without having to enter in all the
commands manually.
I feel that it is probably stable enough to be released now, but I would
On Fri, 2003-11-28 at 12:23, Leif Madsen wrote:
However, the variable names have changed since I posted that. They are
now:
externip ; external ip or FQDN
localnet ; internet ip of asterisk
localmask ; subnet mask of internal machine
I should also note that they've only
Hi David,
I'm currently considering various PBX solutions for our office telephone
network, and would very much like to use Asterisk. Currently, my
research is incomplete. I have been recommended to use the above cards,
but it is unclear from my Googling whether my configuration will work:
Hello,
I have a problem. When Idial to asterisk with
H323 I do not hear ringing applecation (phone rings but i do not hear ringing
tone in handset). I have tried with Cisco 2600 H323 and Quintum
H323.
But when I connect I can hear ringing appl. What
can be wrong? Configuration is wrong?
ranga wrote:
I agree with you. But the issue is, how could I fix the domain name
variable? This should not be static. The target domain changes as per the
choice of the user that is connected through softphone. For example, you are
connected to edvina.net. Now I want to call you from my
I'd like to put up on the wiki the known working
3.3v MotherBoards people are using in production...
I am very interested w/ppl with dual te410's with lots of concurrent
channels in use
Please dont post just your fav spec boards JUST ppl with working stable
installs with TE410s, if possible with
see /path/to/asterisk/channels/h323/README and then
/path/to/asterisk/channels/h323/h323.conf.sample
Jeremy McNamara
César Rico wrote:
Hi, this is Cesar Rico
I'm a new Asterisk user, I would like to know how can I develop an
application with voice over IP on H323 protocol, I've read
- Original Message -
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 29, 2003 3:34 AM
Subject: Re: [Asterisk-Users] call waiting disable in sip
what would happend if all operators are busy? would app_queue exit?
would it schedule the call to
Michael,
Thanks a bunch, I downloaded from inaccessnetworks.com thinking that it is
the latest :). Ok I will upgrade it. just for the record, following worked.
exten = _87.,1,Dial(OH323/H323:${EXTEN:[EMAIL PROTECTED])
Cheers
Sathya
Date: Fri, 28 Nov 2003 11:28:59 +0200
From: Michael Manousos
if you have 80calls going, its time to think on getting a good dedicated
server, switches, for the work and UPS with big batterys also some good
power supplie:)
Miguel
On Fri, 2003-11-28 at 12:51, Matthew Asham wrote:
Hello,
I'm trying to figure out what portions of Asterisk need a lot of CPU
Hello,
Is there anyone out there who is using asterisk as a VoIP Gateway to a
Fujitsu 9600? We have the existing system in place, and I have a mini
gateway functioning using a devel kit from digium. I am a systems admin,
and know near nothing about the Fujitsu, and could really use some
Message: 9
From: Leif Madsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Organization: http://www.hacklocalhost.com
Date: 27 Nov 2003 23:10:42 -0500
Subject: [Asterisk-Users] Asterisk behind NAT How to do it.
Reply-To: [EMAIL PROTECTED]
Thanks to ww and his patch on bug #104, I have
I have tried everything and still can't place / receive calls from the fwd network.
At one point today I was able to call my test machine on the fwd network, I'd answer
the call on the test machine (which stated Call Connected), but then the computer I
was calling from, through the Asterisk
Anyone else having timeout problems with IAXtel? Here's the logfile output,
user names, passwords, and destination phone numbers have been changed to
protect the guilty
-- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1,
IAX/someuser:[EMAIL PROTECTED]/[EMAIL
On Fri, 28 Nov 2003, Steve Rodgers wrote:
Anyone else having timeout problems with IAXtel? Here's the logfile output,
user names, passwords, and destination phone numbers have been changed to
protect the guilty
I just called myself. It worked fine.
--
Joel
Strange,
It's back up for me as well...
On Friday 28 November 2003 18:49, Joel Maslak wrote:
On Fri, 28 Nov 2003, Steve Rodgers wrote:
Anyone else having timeout problems with IAXtel? Here's the logfile
output, user names, passwords, and destination phone numbers have been
changed to
Hi,
I have a deltathree account and I can place calls
but I can't receive calls. I use Grandstram sip phones. When I call my
deltathree phone # the voicemail is answer :((
I need some help and solutions from the guys who
allready are using deltathree. I search on Internet and I try all
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