Surely you mean grammar? Sorry, I just had to point that out :)
Personally, I'd take issue with the title -- if you need to do a
small-office setup by-the-book, then chances are you're not
resourceful enough to find the required information online -- and if you
can't even find the basics online,
[I messed up the in-reply-to of this email, sorry]
Kevin P. Fleming wrote:
[EMAIL PROTECTED] wrote:
1. Dial Tone - No, Yes - Precise, Yes - SCC
Not relevant on a PRI.
Actually the dialtone may be optional on a pri. It is mandatory for
EuroISDN but in other implementations it may not be.
I`m currently writing my second asterisk book hoping to cover all the
needs for everyone,
I`m currently looking for a Editor if anyone is interested someone that
would be willing to read over the book recommend ways of fixing it and
making it more user frendly if anyone is interested please
Sorry you feel Strongly about the book but, Iâm not trying to rip anyone
off but as it said on the back of the book and under the author Iâm new,
to writing books I am trying to help the community out, But I guess this
doesnât really matter to you,
But if you know so much about asterisk and the
Ithink the question
isperfectsince VLANs are a great way to provideQoS and havent
really been discussed here (at least lately).
Can you be more specific as to your problem?
Did you set vlan tagging on the phone? Did you trunk yourswitch
portsall the way back to the router (ISL or 801.q,
What solution provides a higher number of simultaneous calls?
I found this http://www.mera-voip.com/voip/sip-hit.php.
They claim 150 with a dedicated server and relatively modest hardware.
Thanks,
Steve Totaro
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL
Just a word of advice. This should not be posted to the list. Even the
subject is addressed to one person, so why send it to the entire list. It
certainly does not make you look very professional.
I can appreciate your desire to be first to the market but common sense
would dictate that you
I found my error: the TDM01B (1-port FXO TDM400P bundle) ships with
the single FXO module in position 4, not position 1. Thus using fxsks=4
in zaptel.conf and channel = 4 in zapata.conf fixed things.
- Mike
*CLI -- Executing Dial(SIP/555-83ee, ZAP/1/92262802) in new stack
Jul 23 13:50:24
[EMAIL PROTECTED] wrote:
(B
(B if anyone would like to have there name under the
(B Contractors list please email [EMAIL PROTECTED]
(B
(Band why don't you just list the entries on the Asterisk
(BConsultants Wiki page at voip-info.org?
(B
(BI`m hoping to have this next book, ready to suite
Hello,
Ive recently purchased Adit 600 with 3FXS and 1FXO
to be connected to my * server via T100P card. This is the output of status
equipment command in the Adit600:
For some reason the FXO card is seen as FXS, why? Is it ok?
On the card it is written FXO.
Regards,
Shlomi
Hi all,
I´m using X-Lite as
SoftPhone in Asterisk. I have configured this:
[101]
;Turn off silence
suppression in X-Lite (Transmit Silence=YES)!
;Note that Xlite sends
NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=jozeph
callerid=Jozeph
Brasil 5678
Here is the output of the status equipment command
:
BootCode Version: 1.23
CardType Status SW Vers CLEI
-- ---
SLOT A T1x2 Present 6.1.2 SIC3DH0CAA
SLOT 1 FXSx8 Present 1.09 SIC3GJ0CAA
SLOT 2 FXSx8 Present 1.09 SIC3GJ0CAA
SLOT 3 FXS5Gx8 Present 1.10
Shlomi Bachar wrote:
Ive recently purchased Adit 600 with 3FXS and 1FXO to be connected to
my * server via T100P card. This is the output of status equipment
command in the Adit600:
For some reason the FXO card is seen as FXS, why? Is it ok? On the card
it is written FXO.
I would ask ADIT,
Shoot the questions to me offline if you'd like...
--
W. Kevin Hunt
CCIE #11841
www.huntbrothers.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Sent: Saturday, July 24, 2004 10:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
why offline? this is good info for the archives.
- Original Message -
From: W. Kevin Hunt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 25, 2004 10:25 AM
Subject: RE: [Asterisk-Users] Layer 3 VPN Question
Shoot the questions to me offline if you'd like...
--
W. Kevin
Sorry, there seems to be an above average amount of squaking about
either where to post answers (top or bottom posting) and when to take it
offline. I'll be happy to answer online now that at least 2 people are
interested...
As far as the question, what is the exact setup you are attempting to
On Sun, 25 Jul 2004, Jay Milk wrote:
Surely you mean grammar? Sorry, I just had to point that out :)
Personally, I'd take issue with the title -- if you need to do a
small-office setup by-the-book, then chances are you're not
resourceful enough to find the required information online --
What are the criteria you use to select which DB to use with *
Built in DB1/Berkley DB
MySQL add in
Postgres
Unix odbc
Brian's dbodbc
Beyond just having a relational DB or not.
Performance?
DB size?
Ease of Access?
Portability?
Gui/browser access
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Hash: SHA1
On Saturday 24 July 2004 12:37 pm, Osvaldo Mundim Junior wrote:
All right Steve. I'll ask them..
But if anybody knows that, please post an answer to the list. This is a
very important Asterisk security configuration to avoid people call you
On Sunday 25 July 2004 11:24, Frank wrote:
What are the criteria you use to select which DB to use with *
Built in DB1/Berkley DB
MySQL add in
Postgres
Unix odbc
Brian's dbodbc
If you don't need the relational aspects I'd probably use DB1/Berkely -- it's
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Saturday 24 July 2004 08:29 am, Jozeph Brasil wrote:
I do that. But when I play MusicOnHold the music is played slowly! I don´t
know why... but is how the bitrate is playing with a different number.
You need to have a bitrate of 8000, and in
Doh! The reason it changed when I upgraded is because I was compiling VM
with Mysql, and I had the mailbox definitions in the voicemail.conf
flat-file.
I put the definition in the SQL database and it works fine, now. :-/
thanks for kicking me into the right direction :)
yours,
matthew
Are
Very welcome,
Glad to have helped.
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matthew
Simpson
Sent: 25 July 2004 17:46
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] hang up when going to voicemail
Doh! The reason it changed when I
I have googled and searched the wiki with no luck.
Is there a way to get * to report on trunk line utilization.
Like a busy report or a usage histogram?
Hi,
Please keep this discussion on-list. I did search the list 3 weeks ago
on not much usable did show up.
Here is my scenario, fyi.
I have sip/iax phones registered on my * server.
My ISP can also do A-Z termination and provide local did numbers and
controls Qos via VLAN/CoS.
If I use the
I read the wiki and looked at the examples, but I'm
still having problems. I have a Digium 4 port card
with POTS lines plugged into all four ports. How do I
play the congestion tone the the caller when they try
and dial out but all the lines are in use?
should something like this work?
I just started service with Broadvoice.com and everything seems to work.
However, apparently my understanding of incoming sip contexts is less
then what I thought it was. Could someone point me in the right direction?
(* on a public address, CVS-HEAD-07/12/04, C7960 phones)
In my sip.conf I
I am setting up an Asterix server and would like to know what you people
use to administer, I would prefer a web based interface.
Grateful for any suggestions,
Thanks
Benny
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Sun, 25 Jul 2004 11:24:09 -0400, Frank [EMAIL PROTECTED] wrote:
What are the criteria you use to select which DB to use with *
Built in DB1/Berkley DB
MySQL add in
Postgres
Unix odbc
Brian's dbodbc
Beyond just having a relational DB or not.
The problem I'm having with understanding this is for incoming calls
from broadvoice. If I remove the context=from-broadvoice from the
above, incoming calls from broadvoice are dropped into the bogon-calls
context (no service available message).
just add the context = from-broadvoice to the
Hello,
After much searching of voip-info.org and google, I'm finally giving in and asking the list.
The setup I have is this:-
Single X100P card in a Debian system
Inbound/Outbound POTS line connects to the X100P
Sipura 2000 and Budgetone 100 on the LAN
1 Cordless and one conventional phone
Not even sure how important this is considering the state of many of the
online docs...
I have doxygen installed as is noted for the requirements for 'make
progdocs', but the make doesn't find dot. I have no idea where dot went, is
or should have been...
I am installing und Suse 9.0 and it's
The 'dot' program comes with the GraphViz package. You can also edit the
contrib/asterisk-ng-doxygen file and set HAVE_DOT = NO (the default is
YES)
James
On Sun, 25 Jul 2004, Lyle Giese wrote:
Not even sure how important this is considering the state of many of the
online docs...
I have
Hello,
I've been working on an * dialer application, whereby a requirement is that if no one
answers the call, a message must be left on voicemail. I've been using the
record(tmp.gsm) function with silence detection enabled to wait for the greeting to
finish before speaking.
However, on
I had my asterisk configuration working very well with broadvoice, but
it stopped working this afternoon.
I plugged the Cisco 7960 phone I used for my origional signup (just a
few days before they offered generic BYOD) and it works fine. I did
notice it seems to do all of its comunication
Hi guys.
I have a XP100P Clone , and the busydetect dont
work for me..
PSTN---Asterisk---Sip---AsteriskPBX
Any call from pstn side dont disconnect ... I have
no disconnect supervision and busydetect dont work...
Please Help me.
Zapata.conf
It isn't August yet...
At 05:43 PM 8/22/2004, you wrote:
It's
more easy download tarball and compile it.
srsergio
-Mensaje original-
De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] En nombre de Yan
Enviado el: jueves, 22 de julio de 2004 13:31
Para: [EMAIL PROTECTED]
Asunto:
Yes, It is in CVS now.
Duane Cox on 2004/7/23 09:44 wrote:
Thanks, I _finally_ got unixODBC and FreeTDS working with MSSQL. I
hate to through all that hard work out the door, but I like your idea
better.
Is it in cvs now, ready to go? I read that mark was waiting on a fix... ?
Thanks for
First, let me apologize for following up on my own message. I traced
the connection the tftp configured 7960 phone was using and want to see
how close I can make asterisk approximate this for the best reliability.
The phone does an srv lookup for _sip._udp.proxy.broadvoice.com which
returns
I had my asterisk configuration working very well with broadvoice, but
it stopped working this afternoon.
I plugged the Cisco 7960 phone I used for my origional signup (just a
few days before they offered generic BYOD) and it works fine. I did
notice it seems to do all of its
I am having issues also.
I called them and I was told that they are doing upgrades on their network.
Zac
On Mon, 2004-07-26 at 04:55, Charlie Hedlin wrote:
I had my asterisk configuration working very well with broadvoice, but
it stopped working this afternoon.
I plugged the Cisco 7960
On Sun, 2004-07-25 at 04:56, Peter Svensson wrote:
[snip]
I'm not sure of the correct answers to any of these. I do know that I want
to be able to get caller ID and the number dialed for the applicationthat I'm
building.
Most of these questions are for trunks being delivered over a
Hey All,
We are running a small SIP/IAX termination service at the moment
(planning on growing it) with 2 asterisk machines. One terminates the
SIP/IAX calls from our customers and one is our gateway to our upstream
provider. Both machines are logging CDR data to the same postgres table
using the
I called them also. They are upgrading sip.broadvoice.com
I hope it works for them. They seem good so far.
-- David Hickman
Pots 314-865-4752 x1 business x31 home
FWD 23633
IAXTEL 700-865-4752
AOLIM fsckrmrf
ICQ 7059948
Yahoo dhickman
THIS IS INSANE! I THOUGHT TECHNOLOGY
Hello Joe,
Monday, July 26, 2004, 2:33:07 AM, you wrote:
JB I read the wiki and looked at the examples, but I'm
JB still having problems. I have a Digium 4 port card
JB with POTS lines plugged into all four ports. How do I
JB play the congestion tone the the caller when they try
JB and dial out
On Sun, 25 Jul 2004, Rich Adamson wrote:
I just started service with Broadvoice.com and everything seems to work.
However, apparently my understanding of incoming sip contexts is less
then what I thought it was. Could someone point me in the right
direction? (* on a public address,
[EMAIL PROTECTED] (Rich Adamson) writes:
However, for the last hour or so their site has been unreachable with
an icmp destination unreachable coming from 199.232.42.62, which belongs
to Cambridge Entrepreneurial Network in Quincy MA. Would guess either
someone upgrading hardware or a failure
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