- Original Message -
From: Michael A Rowley [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 14, 2004 11:40 PM
Subject: [Asterisk-Users] Data through T1, nethdlc
7) sethdlc hdlc0 hdlc_net
I use sethdlc hdlc# cisco but only because that's how I learned.
Next you have to bring
We manage our own VOIP solution using Asterisk.
Has anyone had success with an IP-PSTN provider? I'm looking for someone to terminate
SIP calls to the PSTN in the Seattle, Washington area. (vice-versa as well if possible)
Yes, I could do it myself via asterisk and digium cards but I would
We provide SIP and H323 origination and termination worldwide.
Extremely competitive rates. We have may Asterisk clents.
Thanks,
Aram Ter-Martirosyan
Senior Account Manager
Hi-Tech Gateway, Inc.
http://www.hi-teck.com http://www.hi-teck.com/
1225 Grand Central Ave.
Glendale, CA 91201
Mike Machado wrote:
I have done a little work on asterisk and database integration. Below is
a link to app_dbmysql, modeled after Brian's app_dbodbc but for pure
MySQL.
I also ported the mysql-vm-routines.h to ODBC in case anyone is
interested.
COuld you please add your apps and changes to the bug
tmpm wrote:
The ones that come to mind are en_ca, where every sentence has an ..eh?
ending.
Our Weasels are eating doughnuts, and drinking beer ..eh?
much like Aussie en_au which prepends a G'day mate.. and ends with the
..eh?
G'day mate, our weasels be puttin our phone system on the
Hi,
Has anyone got the X100P to work with an anlogue line
in the Republic of Ireland?
I have the X100P installed but zttool indicates a Red
Alarm status on the card. It is on its own interrupt
and I have tried different PCI slots but all to no
avail.
Are there any alternatives to the X100P that
Hi,
I am presently thinking of making just a demo application because i need to
assemble it real quickhooking up an analog phone line for an input
channel...with a dial out feature as well..
I think I would need an XP100 wildcard as FXO .not sure of what to use for
FXS... Incidentally steven
Hi Steve,
Check your config in settings:redirection:event?
None
mailbox = [EMAIL PROTECTED] (change extension and context to match yours)
[EMAIL PROTECTED]
can it be it is the firmware ?
have 2.05b.
greeting
nicolas
Steve Totaro wrote:
Check your config in settings:redirection:event?
glad you enjoyed it Ollie...heh..
At 04:04 5/15/2004, you wrote:
tmpm wrote:
The ones that come to mind are en_ca, where every sentence has an ..eh?
ending.
Our Weasels are eating doughnuts, and drinking beer ..eh?
much like Aussie en_au which prepends a G'day mate.. and ends with the
..eh?
Why does voicemail prompt me for an extension instead of just asking my
password?
Because there is no Voicemailbox 99 in that context in
your configuration.
[voice-mail]
exten = 99,1,VoicemailMain([EMAIL PROTECTED])
exten = 99,2,Hangup
In your example, $EXTEN will always be 99, because
Hi,
-Original Message-
Finland, Denmark, Iceland, Sweden, the Netherlands, Belgium,
Brazil, Saudi Arabia, Uruguay,India all use DTMF
So, logically the DTMF solution would be attacked first...
but then I do have a bias..
Add it to:
Since the irc channel wasn't any help, I will try posting my problem here.
I have two TDM400Ps less than a week old in a PC. All of the FXS ports
work fine, and all of the FXO ports worked fine up until thisafternoon. If
I try to dial in, I get a busy signal, if I try to dial out, all I hear
Hi,
I have setup and configured asterisk server using SIP. I defined
two users 2000 and 2001 in the sip.conf file. The extension
defintions from sip.conf are shown below:
[2000]
type=friend
username=2000
secret=hi
host=dynamic
context=from-sip
mailbox=100
[2001]
type=friend
username=2001
Hi,
I use asterisk with a Digium E1 (wct1xxp). On my old server,
everything went fine, but after having built the card to a new one, I
only have problems:
-- Executing Dial([EMAIL PROTECTED]:18308]/1, Zap/1/853)
in new stack
May 15 14:10:37 NOTICE[15376]: app_dial.c:554 dial_exec: Unable
I did everything you specified but couldn't get it to work on 2.4.25.
With tcpdump I could see incoming ICMP, UDP, and TCP packets on each
end, but they would just fall off the end of the earth--no replies at all.
The sethdlc and sethdlc-new programs are in the zaptel directory on CVS.
Hi all,
Did anyone see this problem.
I have two X101P in my asterisk Box.
When i receive a call, everything goes Ok, but after a short period of time
i receive a hangup ..
Sometimes happen at 1:50min others at 05:00min but always happend.
Did anyone see this before ??
How can i make the cards use
On Fri, 2004-05-14 at 15:21, Digvijay Singh wrote:
Hi,
I am presently thinking of making just a demo application because i need to
assemble it real quickhooking up an analog phone line for an input
channel...with a dial out feature as well..
I think I would need an XP100 wildcard as FXO
What purpose does echocancelwhenbridged=no/yes have on a tdm04b (or
x100p) fxo interface when all phones are sip 7960's on the same wire?
Rich
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 15, 2004 7:37 AM
Subject: Re: [Asterisk-Users] Data through T1, nethdlc
I did everything you specified but couldn't get it to work on 2.4.25.
With tcpdump I could see incoming ICMP,
Digvijay Singh wrote:
Hi,
I am presently thinking of making just a demo application because i need to
assemble it real quickhooking up an analog phone line for an input
channel...with a dial out feature as well..
I think I would need an XP100 wildcard as FXO .not sure of what to use for
FXS...
On Sat, 2004-05-15 at 07:56, Carlos Arnt wrote:
I have two X101P in my asterisk Box.
When i receive a call, everything goes Ok, but after a short period of
time
i receive a hangup ..
Sometimes happen at 1:50min others at 05:00min but always happend.
Did anyone see this before ??
Try,
http://www.IPKall.com http://www.ipkall.com/
Arick
-Original Message-
From: Chad Brown [mailto:[EMAIL PROTECTED] On Behalf Of
Chad Brown
Sent: Friday, May 14, 2004 11:27 PM
To: [EMAIL PROTECTED]
Subject: IP-PSTN / PSTN-IP Gateway Service Providers
We manage our own VOIP
you also have
fromuser
fromdomain
use em
bkw
- Original Message -
From: usmankhan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 15, 2004 6:10 AM
Subject: [Asterisk-Users] TO header field bug using asterisk
Hi,
I have setup and configured asterisk server using SIP.
On Fri, 2004-05-14 at 13:58, Scott Stingel wrote:
I suggest:
* Use 1 processor (example: 2.8Ghz P4) for every 4 E1's (example: one
TE405P card) when you have that much call setup traffic.
Scott,
On my own testing I could get 200 simultaneous calls on a Xeon 2.4
using SIPP, each call
Any clues as to what this is? Google isn't much help. Both cards have the
power connector plugged in, and both randomly give me static instead of a
dialtone, or an outside line.
Cards are 1 4 Port TDM400 FXO, and 1 4 Port TDM400 FXS. Computer is a
Compaq Proliant ML330
The full error is as
On Sat, 2004-05-15 at 03:17, Aaron Clauson wrote:
I have the X100P installed but zttool indicates a Red
Alarm status on the card. It is on its own interrupt
and I have tried different PCI slots but all to no
avail.
If you plug a regular ANALOF phone into the second port on the X100P do
you
I've gotten several Power alarm on module 1, resetting since I
installed a quad FXS TDM400 card. Dell 400sc.
Does your motherboard have the A-B-C-D LEDS above the keyboard/mouse
connectors?
[EMAIL PROTECTED] wrote:
Any clues as to what this is? Google isn't much help. Both cards have the
On Fri, 2004-05-14 at 13:58, Scott Stingel wrote:
I suggest:
* Use 1 processor (example: 2.8Ghz P4) for every 4 E1's (example: one
TE405P card) when you have that much call setup traffic.
Scott,
On my own testing I could get 200 simultaneous calls on a Xeon 2.4 using
SIPP,
each call was
Hi
i have buy two license of G729 codec, but after run Registration
program i notice this error
Registration unsuccessful... Error code: 110
ERROR!
Your Internet connection is probably behind a proxy and the
Registration program can't communicate with our server,
however
Hi Thomas,
Check your /etc/zaptel.conf and /etc/asterisk/zapata.conf. Probably you
have not configured your channels at zapata.conf.
Daniel
Thomas Schroeter wrote:
Hi,
I use asterisk with a Digium E1 (wct1xxp). On my old server,
everything went fine, but after having built the card to a new
How do you setup * so that a user can transfer a call
directly into someones voicemail?
Damian
From: Damian Dicks
How do you setup * so that a user can transfer a
call directly into someone's voicemail?
There are a few ways, but the way I have it:
All our extensions are four digits and start with 7, so I have the system
setup so that if you transfer a call to *EXTN (eg: *7202) the call
Hi,
I suspected that I the analogue phone should have got
a pass through signal when the power was off to the
server, unfortunately it doesn't. I kept asking digium
support about that but they didn't give me an answer.
The problem is how do I identify whether the X100P is
incompatibel with the
On Sat, 2004-05-15 at 12:01 -0700, Aaron Clauson wrote:
Hi,
I suspected that I the analogue phone should have got
a pass through signal when the power was off to the
server, unfortunately it doesn't. I kept asking digium
support about that but they didn't give me an answer.
The problem
On Sat, 2004-05-15 at 14:01, Aaron Clauson wrote:
I suspected that I the analogue phone should have got
a pass through signal when the power was off to the
server, unfortunately it doesn't. I kept asking digium
support about that but they didn't give me an answer.
The problem is how do I
Same hapened to me, seems that authentication server is not working.
Hi
i have buy two license of G729 codec, but after run Registration
program i notice this error
Registration unsuccessful... Error code: 110
ERROR!
Your Internet connection is probably behind a proxy and
Title: Some doc
I've posted a couple of docs that are intended to be helpful at http://www.rainiernetworks.com/29535/33701.html
1. A case study of one of our customers using a fairly straightforward T1 coming in to the Asterisk box
2. The installation steps we followed
3. The conf
On Fri, 14 May 2004 05:08:54 -0500, Simon Dorfman wrote:
On 5/14/04 4:19 AM, Robinson Tim-W10277 [EMAIL PROTECTED] wrote:
There should probably be en_uk, en_us, en_ca, en_za, en_nz, en_oz, en_ie and
en_in etc to allow each English-speaking country to localise prompts.
To further complicate
FWIW, I ordered one of the beta units from Voxilla last month. They
notified me today that the unit has shipped. I'm still looking for a
good FXO i/f for my two remaining PSTN lines. Should be interesting to
see if they work as well as the SPA-2000.
Michael
--
Michael Graves
I was having the same problem as well as few other users that I know of.
The registration issue seems corrected but we are having problems with
our customers not receiving ring back tone when calling us.
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: Friday, May
What version of Asterisk are you using. 0.9.0 had issues with ringback
in some situations. One of these was parallel destinations on a Dial
line like Dial(SIP/123SIP/678). Use CVS -stable or if you like to live
life in the fast lane, CVS -head.
On Sat, 2004-05-15 at 19:14, Kevin wrote:
I was
H - I do remember some discussion way back, but I never had any issues
to speak of myself. I've got a fairly small (but world-wide since I got
offices in Denmark and in Malaysia) setup with 3 asterisk boxes each running
a bunch of local ISDN lines (using Fritz cards) and a mix of Snom,
does anonybody know what is the status of gsm codec in snom phones ?
they were some issuses in archives, some problems so i would like to
know what is the actual status.
best regards
Marian
I had problems with the SNOM phones with GSM when they first came out. One
of the first couple
Well -
the only problem I had with my Snom 200 was that after the update it mentioned
that a new version was available - which was in fact an old version. I
just disabled the automatic update and then no problem.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Hi all,
Is it possible to have one asterisk box controlling 110 extensions using commom
telephones.
In a first moment, I am not going to connect to PTSN.
--- |- # 1
| * | |- # 2
| box |-|- # ...
| | |- # 109
--- |- # 110
If you use SIP or 2xTE4XX card+channelbanks... you can scale into alot
higher numbers of extensions.
bkw
- Original Message -
From: Alex G Robertson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: Alex Robertson [EMAIL PROTECTED]
Sent: Saturday, May 15, 2004 8:43 PM
Subject:
Eric,
Thanks for the reply. Everything had been working fine for quite some
time till the other day. My other inter-machine IAX connections work
perfectly fine as well as other origination connections.
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: Saturday, May
Kevin wrote:
Thanks for the reply. Everything had been working fine for quite some
time till the other day. My other inter-machine IAX connections work
perfectly fine as well as other origination connections.
I have not updated Asterisk on our DID machines in over 3 weeks. I
cannot reproduce
Steve Totaro wrote:
2600mhz cpn crunch whistle? bump the oper off the line?
Holy crap Batman! You've got a whistle that does 2.6Ghz? Perhaps you
should look into some RF exposure safety literature.
Hz = cycles of function per second (function being signwave, sawtooth,
squarewave, etc)
KHz =
Quoting Jim Rosenberg [EMAIL PROTECTED]:
Let's suppose you have a PSTN line with multiple extensions. One of these
extensions is connected to an Asterisk FXO port. For reasons that I won't
go into, the normal configuration of the system is that when a call comes
in on the PSTN line,
Title: *8 (call pickup) using Manager or AGI interfaces?
I'd like to programmatically do the equivalent of a *8 using either the Manager or AGI interfaces or some other Asterisk interface.
In this scenario, a line is ringing, the pickup program executes with an argument that is the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jim Rosenberg
When I hear one of these extensions ringing, I want to be
able to dial an
extension on the Asterisk phone system that will allow the
Asterisk FXO
port to answer, and from
At 7:20 PM -0400 on 5/15/04, Jim Rosenberg wrote:
Let's suppose you have a PSTN line with multiple extensions. One of
these extensions is connected to an Asterisk FXO port. For reasons
that I won't go into, the normal configuration of the system is
that when a call comes in on the PSTN line,
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