Title: IAX2 Problem and Question
Dear Asterisk Users.
I have been setting up IAX between two servers, one in the USA and the other in UK so that I can pass help desk and general calls from one call center to another.
I seem to be having an issue. When I set up IAX between my two servers I g
Hi,
I just got FWD and ICH behind a Linux NAT firewall working. The funny
thing is that I cannot get them to _both_ work at the same time. Its
either one or the other.
The setup is using a local Sipura and the standard extension setup
(ICH=6, FWD=7) as documented on some sample * configs. The
Hi,
I restarted my * box before and now my second X100P card has stopped
showing up in Asterisk. It shows up in my dmesg in Gentoo, but only 1
channel is being registered:
[chan_zap.so] => (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXS
I thought I saw online in a list somewhere
that there’s an improved Call Queuing app out; supposedly it has the capability
to tell the caller how far down the queue they are, etc? I saw one post about it
somewhere but then no mention of it anywhere else…
Andy, [EMAIL PROTECTED]
No, the voice mail runs without any
hardware.
Check out http://www.voip-info.org/ for information
about implementing voicemail into Asterisk.
- Sam
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
PTCHENSent: Sunday, April 04, 2004 8:18 PMTo:
[EMAIL PROTECTED]Subject: [A
Hi,
I am a Asterisk beginner, and are going to use Asterisk
PBX voicemail service.
Is it necessary for me to get any hardware to run this service? Thanks
!
Hi...
Actually, I have to digium cards install on my
server running asterisk. They are X101P and TDM400P. Right now, X101P is running
perfectly, but the another card doesn't work. In fact, it is not reconized by
the server.
When I try to load wcfxs.o it shows me the
following message:
We have a system with multiple Asterisk gateways and an SER
proxy. I was wondering if people could share what they have been doing for
prepaid billing on either multiple Asterisk gateways, or a single SER proxy?
Do people use a radius or do they go right from the mysql
database? I ha
I have used curl to reboot the GS101 as follow:
curl -c cookies.txt -d"P2=x&Login=Login&gnkey=0b82"
http://192.168.1.xxx/dologin.htm
curl -b cookies.txt http://192.168.1.xxx/rs.htm
Put these 2 lines in a script and use cron to reboot everyday.
--
David Kwok
Tel: 612 99292086 ext 1002
Iaxtel
Title: Silence suppression on SIP calls generated from Asterisk?
Let's say that I have a call coming in to Asterisk through a TDM400P and going out through SIP to someone on the Internet. Is there any configuration option that would allow me to do silence suppression on the RTP stream genera
On 2004 Apr 04, at 15:51, Aaron Martin wrote:
Is there anyway to display parked calls or call queues on this, so that
users can just drag and drop to connect to the calls?
Is there anyway the developer of this tool could create a separate
mailing
list for this and refer all questions there? An a
On Mon, 5 Apr 2004, Leo Ann Boon wrote:
> I'm trying to add custom billing info into the CDR records. I did a
> SetCDRUserField from my agi. Asterisk seems to acknowledge the call, but
> the value is not anywhere in the CDR record. I checked the CSV and Mysql
> CDR table. The field is always blank.
Hi all,
I'm trying to add custom billing info into the CDR records. I did a
SetCDRUserField from my agi. Asterisk seems to acknowledge the call, but
the value is not anywhere in the CDR record. I checked the CSV and Mysql
CDR table. The field is always blank.
Anyone had any success with this?
[EMAIL PROTECTED] wrote:
> I know that you can reboot the GS phones by hitting the
> rs.htm URL on the phone. But, you have to log in to the web
> interface before doing this.
I've attached a php script (quick and dirty hack) that resets the
specified Grandstream devices. It requires the Snoopy
Is there anyway to display parked calls or call queues on this, so that
users can just drag and drop to connect to the calls?
- Original Message -
From: "Nicolas Gudino" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, April 02, 2004 8:52 AM
Subject: [Asterisk-Users] ANNOUNCE: Fl
Hi Guys,
My name is Marcias, and I am setting up for the
first time an Asterisk PBX, I am learning as I go along. I have been able to
download and install Asterisk, Libpri and I have been able to get Asterisk up
and running. I have several questions:
1 .I can call the Asterisk server from
Larry Keyes wrote:
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below. Any
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below. Anyone know what is
goi
Robinson Tim-W10277 wrote:
Use RC16. This seems to solve our issues on a UK ISDN2e line.
Rgds
Tim
Thanks for the tip - rc16 didn't work but rc17 magically fixed things. I
also had to ensure that the dip switches on my by highway box were set
to "S" and "IN"
regards,
Julien
__
On Sun, 4 Apr 2004, Serge Mankovski wrote:
> Hi
> I am trying Manager interface for originate a call. This is what I get
> ---
> Action: Originate
> Exten: 555
> CallerID: test <6656>
> Context: local
> Timeout: 600
> Channel: SIP/8782
> Priority: 1
>
>
> Response: Error
> Message:
Agreed...not that there is much of a security risk with pif files when you
don't execute them...
It was just a passing comment as I am on about 10-15 mailing lists and this
one is the only one I receive viruses from...and this is the 2nd or 3rd
time...all the other lists I am on don't allow attach
I contracted with Digium for this enhancement and am waiting for it to be
completed.
Tilghman Lesher wrote:
On 2004 Apr 02, at 12:04, Brian Capouch wrote:
I don't want to re-invent the wheel if someone has already
hacked a way to do this.
One of my customers has a number of stores, and he wants
On Sun, Apr 04, 2004 at 01:00:04PM -0400, Andrew Kohlsmith wrote:
> > Can't we get the mailing list to not allow mails with .pif files attached?
>
> Can't we teach posters not to hit reply, erase the subject and content and
> start a completely new thread, screwing up threading for everyone else?
G'day,
If you rely on the server side to implement security for you then you've already lost.
You can filter on your smtp server end, or your mail client end.. this will be far
more reliable.
Second hint for you, as opposed to dropping 'problem' extensions, why don't you
whitelist the extensio
Sorry, the only reason the content was missing is that I deleted the email
when it had a virus...if you would prefer, I'm sure someone could send it to
you...
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, April 05, 2004 5:00 AM
Sub
> Can't we get the mailing list to not allow mails with .pif files attached?
Can't we teach posters not to hit reply, erase the subject and content and
start a completely new thread, screwing up threading for everyone else?
FWIW, I do agree that there should be no attachments allowed to this lis
Darren Sessions wrote:
I’ll apologize right away for asking stupid questions. J
System Setup:
SER = Proxy
Asterisk = Voicemail
All sip based setup.
1. What Is required to make asterisk –NOT- accept inbound
calls/signaling from an unknown host? I tried the peers in
sip.conf but
Can't we get the mailing list to not allow mails with .pif files
attached?Matt Riddell
Hello,
I tried to find out,
when a domain name is checked when I use a domain name instead of a IP Address
for externip in the sip.conf but I did found something on that. Does someone
knows the behavior on that?
My problem is, that
I have a dynamic IP address outside. This IP-address will
I’ll apologize right away for asking stupid questions.
J
System Setup:
SER = Proxy
Asterisk = Voicemail
All sip based setup.
What Is required to make
asterisk –NOT- accept inbound calls/signaling from an unknown host?
I tried the peers in sip.conf but it stil
Ryan Parlee wrote:
Okay, but if I do
/usr/sbin/asterisk
Then when I connect, using -r I don't get debugging information. Isn't
there a way to make asterisk deamonized and still get the - stuff when
you reconnect?
How about spending a few hours studying the WIKI:
http://www.voip-info.o
Hi John,
I found a very basic Betabrite package for linux. It's
nowhere near as clever as the Winwows one but I guess
it could be scripted.
As for the manager interface; it seems that all folks
are interested in there is a phantom operators console
or adding extensions etc.
Are we too far ahead
Hi
I am trying Manager interface for originate a call. This is what I get
---
Action: Originate
Exten: 555
CallerID: test <6656>
Context: local
Timeout: 600
Channel: SIP/8782
Priority: 1
Response: Error
Message: Originate failed
What do I do wrong?
Thank you
Serge
On 2004 Apr 02, at 12:04, Brian Capouch wrote:
I don't want to re-invent the wheel if someone has already
hacked a way to do this.
One of my customers has a number of stores, and he wants
to leave one voicemail that would be delivered to all the
managers at once. Each has a voicemail account on h
On 2004 Apr 03, at 22:55, Sean Cheesman wrote:
Ryan Parlee wrote:
Okay, but if I do
/usr/sbin/asterisk
Then when I connect, using -r I don't get debugging information.
Isn't
there a way to make asterisk deamonized and still get the - stuff
when you reconnect?
just do -vvvr
Actually, -v
Greetings,
I had been working on Asterisk (http://asteriskpbx.org) about 2 years ago .
http://www.marko.net/asterisk/archives/0210/0107.html
Last night with the help of Jesse's rt-soap-server.pl (and some prodding)
I implemented a much cleaner, more repeatable * -> RT phone gateway wit
Hi all
How I configure the Asterisk to work in set with another Asterisk?
I want to balance my customers in some computers with the Asterisk
rounding.
Thank You,
Joao Carlos Moura
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.
HI
If I understand correctly, you are talking about a production of RSS feed
(see http://www.xml.com/lpt/a/2002/12/18/dive-into-xml.html)
I am writing a bunch of java classes that will expose Manager interface in
more readable form (form Java point of view). I might think of writing an
RSS feed
My asterisk is located on 100mbit switched network (HP procurve 2524
switches); on asterisk server itself there is no sound hardware. I have
installed single hisax ISDN BRI adapter which is connected to my PBX.
As for clients, I use mediatrix FXS units with SIP protocol.
One of the mediatrix unit
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