Well...Not sure whats causing it, and I haven't gotten to the box yet to
look at it, but this is the second time my newly built * box has just hung
without warnign or explanation. No IRQ sharing going on (i stole IRQ 3 and
4 after disabling the Com ports) with the TDM400P and T100P. Runs
Ah Sorry, * v 1.0.1, zaptel, libpri, also 1.0.
The box is an AMD XP1800+ w/ 1.5Gig RAM and a ASUS A7V266-E mainboard.
Like I said, previously solid performer.
IDE based drives...
This isnt' the permanent home, that hardware arrived earlier today while I
was out, I know it's a Supermicro
BIX refers to the type of punchdown block invented by Nortel and later
sold to Nordx (BIX is part of the IBDN family). It is functionally the
same concept as 110 blocks, although the two are not compatible.
I guess the closest thing to a breakout box for BIX would be the BIX36,
which allows you
On Tue, 12 Oct 2004 13:19:31 +1000, Adam Goryachev
[EMAIL PROTECTED] wrote:
Well, IMHO, I would expect it perfectly reasonable for one of three
responses from the 'user' community (in order of likelihood):
1) A resounding non-response
2) A response of Well, get a X101P or TDMx0P and try it
On Mon, 11 Oct 2004 18:16:14 -0500, Steven Critchfield
[EMAIL PROTECTED] wrote:
2 cards is the highest number recommended.
That's not entirely correct.
It should be
2 cards is the higher number recommended ***on x86 hardware***
If you use a Mac running LinuxPPC, you can use as many cards as
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Goryachev
Sent: 12 October 2004 04:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards
Wow, that's a really sucky
On Mon, 11 Oct 2004 23:58:30 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote:
what I've read leads me to believe that it won't be
feasible to have large numbers (specifically, many more than two) of
the X100P's in a single system.
The limitation you mention stems from interrupt conflicts which
On Tue, 12 Oct 2004 09:05:02 +0100, Karl Dyson [EMAIL PROTECTED] wrote:
It works most of the time, but drops calls occasionally.
and:
To resolve my problem, I may buy a Sipura-3000
I don't know how much effort you have already made but keep in mind
that there are parameters in the Zaptel
On Tue, 12 Oct 2004 08:15:11 +0300, Lubomir Christov
[EMAIL PROTECTED] wrote:
If you decide to start testing with LineJacks here are some problems
which you will find soon :)
I think it is about time to consider removing those legacy modules
from the main Asterisk distribution altogether for
Hello,
I'm building a 120 IVR server, how can I configure
redundance or failover ?
With 2 servers, mysql and 1 TE410P by server.
I need to share the mysql database and some files.
Mysql cluster look like wery interesting but in alpha
release :-(
Any links or experiences ?
Thanks
bagattin jerome wrote:
Hello,
I'm building a 120 IVR server, how can I configure
redundance or failover ?
With 2 servers, mysql and 1 TE410P by server.
I need to share the mysql database and some files.
Mysql cluster look like wery interesting but in alpha
release :-(
Any links or experiences ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Benjamin on Asterisk Mailing Lists
Sent: 12 October 2004 09:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards
On Tue, 12 Oct
Good day all
We have a pbx system running sip and sipphone(Bughtone)
My question is.If a user is not at their desk,how do I tell it if a call
comes in it should direct it to someone else
Do I need a different phone for this?The only other way is that they
have to switch it off and in my dialplan
On Tue, 12 Oct 2004, Altus Syman wrote:
Good day all
We have a pbx system running sip and sipphone(Bughtone)
My question is.If a user is not at their desk,how do I tell it if a call
comes in it should direct it to someone else
Do I need a different phone for this?The only other way is
On Tue, 2004-10-12 at 11:20 +0200, Altus Syman wrote:
Good day all
We have a pbx system running sip and sipphone(Bughtone)
My question is.If a user is not at their desk,how do I tell it if a call
comes in it should direct it to someone else
Do I need a different phone for this?The only other
It almost sounds like you need to take advantage of using agents in
asterisk. Search the wiki for agent
http://www.voip-info.org/tiki-searchresults.php?words=agentwhere=pages
Thanks,
Steve
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Altus Syman wrote:
Good day all
We have a pbx system running sip and sipphone(Bughtone)
My question is.If a user is not at their desk,how do I tell it if a call
comes in it should direct it to someone else
Do I need a different phone for this?The only other way is that they
have to switch it off
Hey,
I've got this phone, and it's working allmost perfectly ;p
Firstly I get some unknown codec errors in asterisk so the first half
second when I make a call is blurry, after that it's just fine.
Then it's that menu selections won't work.
According to
Hello,
Are you store voicemail and sip clients in sql
databases ?
I 'm looking for sharing files in database
Harry
--- el Flynn [EMAIL PROTECTED] a écrit :
bagattin jerome wrote:
Hello,
I'm building a 120 IVR server, how can I configure
redundance or failover ?
With 2 servers,
Hi Matt,
On Tue, 12 Oct 2004 22:10:24 +1300, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Seeing as this is off topic I thought I'd post to you directly.
I don't think it's off-topic because I always get a lot of email from
Asterisk users who want to know more about how to turn an old Mac into
an
What is the configuration of the TDM400?
Port 1 -
Port 2 -
Port 3 -
Port 4 -
--
-M
There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Hi,
my favorite alternative to cisco 7912G/7940G is Intracom's
Netphone
http://www.intracom.com/en/products/terminal_equip/netphone.htm
today, we use cisco, but we also looking for alternative and
this phone seems to be very well :o)
in-line power
integrated switch (I don't know if supports
Got the same problem on a linux redhat 7.3 box
But everything hangs
Its also runs mail
You can ping it but cant connect on any port,not eve the gui
Michael George wrote:
What is the configuration of the TDM400?
Port 1 -
Port 2 -
Port 3 -
Port 4 -
On Tue, 2004-10-12 at 04:11, Barton Hodges wrote:
Trying again with a different subject...
Currently, if I briefly press the flash hook on my phone, the caller
is placed on hold. I would like for the channel to hangup if I do
this instead, never placing a caller on hold (I'll be using
I'm new to Asterisk.
How big can be sip.conf (and other: iax.conf, extensions.conf...)
Is there point when I must use DB (MySQL...) instead of pure .conf?
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My configuration are:
AlcatelOmni PCX ßà1st
Asterisk Server with ZapCard ßàIAX
trunk over Internetß 2nd
Asterisk Server ßà
SIP phone
I have problem with echo in this configuration. But
when I use sip phone or call trough BRI even if I use IAX trunk I have no
problem
Can someone
I'm stumped.
I'm trying to configure my Asterisk setup to use a new embryonic UK ITSP
(who wants to remain nameless for the moment ...).
They supply their own UA, which works fine.
But Asterisk is proving to be a problem. Ethereal traces show that their UA
is sending SIP packets direct to an IP
Then you use just username.
I don't think the MD5 is critical - the auth=username:[EMAIL PROTECTED] is
what chan_sip2 uses.
You must not use port=5060 in sip.conf - it has to be port=5052. You also
have to do funny things with the externip - point it to the *internal*
address and use nat=yes as
Am Dienstag, 12. Oktober 2004 14:04 schrieb Goran Dj:
I'm new to Asterisk.
How big can be sip.conf (and other: iax.conf, extensions.conf...)
Is there point when I must use DB (MySQL...) instead of pure .conf?
There is no real answer to this question.
Databases are always a good choice as the
Thanks Jeremy !!
We met at the conference. I will do as you say ! ;-)
Regards.
Francisco
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Good day all
This is most likely a new topic but I'm searching for some billing
software for asterisk ,free, if can
I looked at http://www.voip-info.org/wiki-Asterisk+billing and the whole
of the cdr thing in my /var/log/asterisk
Its all all comma separated and I'm sure It cant be that hard to
Thanks for your advice. I will consider moving on.
Regards,
Frank
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On 10/12/2004, Altus Syman [EMAIL PROTECTED] wrote:
Good day all
This is most likely a new topic but I'm searching for some billing
software for asterisk ,free, if can
I looked at http://www.voip-info.org/wiki-Asterisk+billing and the whole
of the cdr thing in my /var/log/asterisk
If you look at
gnophone doesn't want to compile:
/usr/include/linux/ixjuser.h:353: error: syntax error before '*'
token
Have a look at this:-
http://lists.digium.com/pipermail/asterisk-users/2004-August/059263.html
Cheers,
Jon
___
Asterisk-Users
Dave Cotton wrote:
On Tue, 2004-10-12 at 11:20 +0200, Altus Syman wrote:
Good day all
We have a pbx system running sip and sipphone(Bughtone)
My question is.If a user is not at their desk,how do I tell it if a call
comes in it should direct it to someone else
Do I need a different phone for
my bad,wiil have a look
Flynn wrote:
On 10/12/2004, "Altus Syman" [EMAIL PROTECTED] wrote:
Good day all
This is most likely a new topic but I'm searching for some billing
software for asterisk ,free, if can
I looked at http://www.voip-info.org/wiki-Asterisk+billing and the whole
my bad, none of those applications really work for us
and they are too rudimentary of any value.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Altus
SymanSent: Tuesday, October 12, 2004 9:08 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
So what do you suggest
Kanuri, Seshu (Company IT) wrote:
my bad, none of those applications
really work for us and they are too rudimentary of any value.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Altus
Syman
Sent: Tuesday,
OK, I tried compiling from tarball.
I got the zaptel-1.0.0.tar.gz. I untarred it. In that directory where the
tarball unpacked I just duid 'make linux26' (without doing a ./configure
or any changes). Zaptel started compiling but gave the same errors about
crc errors:
*** Warning: zt_ec_chunk
hi
with silence suppression enabled I get these:
Oct 12 15:45:55 NOTICE[1104014256]: rtp.c:289 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
is rfc3389 support planned?
roy
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[EMAIL
Remco Barende wrote:
OK, I tried compiling from tarball.
I got the zaptel-1.0.0.tar.gz. I untarred it. In that directory where
the tarball unpacked I just duid 'make linux26' (without doing a
./configure or any changes). Zaptel started compiling but gave the same
errors about crc errors:
***
Roy Sigurd Karlsbakk wrote:
hi
with silence suppression enabled I get these:
Oct 12 15:45:55 NOTICE[1104014256]: rtp.c:289 process_rfc3389: RFC3389
support incomplete. Turn off on client if possible
is rfc3389 support planned?
I don't know if it's planned, but one of the features required to
hi
http://voip-info.org/tiki-index.php?page=Low%20Bandwidth%20VOIP tells
me it's possible to do = 10kbps (or that someone thinks so), although
http://voip-info.org/tiki-index.php?page=Bandwidth%20consumption tells
me the lowest possible would be G.723.1 at 5.3kbps, resulting in a
total of
On Mon, 11 Oct 2004 23:58:30 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote:
what I've read leads me to believe that it won't be
feasible to have large numbers (specifically, many more than two) of
the X100P's in a single system.
The limitation you mention stems from interrupt conflicts
Why not use an NTP timing source - go stratum 2 or 3. That should be
plenty for a stable clock source.
On Oct 12, 2004, at 9:52 AM, Eric Wieling wrote:
Roy Sigurd Karlsbakk wrote:
hi
with silence suppression enabled I get these:
Oct 12 15:45:55 NOTICE[1104014256]: rtp.c:289 process_rfc3389:
Darren Sessions wrote:
Why not use an NTP timing source - go stratum 2 or 3. That should be
plenty for a stable clock source.
*Timing* is what is needed, not _time_. Two different things. Besides
the obvious problems with using a remote network resource as a timing
device, I don't think many
On Tue, 12 Oct 2004, Christopher L. Wade wrote:
Remco Barende wrote:
OK, I tried compiling from tarball.
I got the zaptel-1.0.0.tar.gz. I untarred it. In that directory where the
tarball unpacked I just duid 'make linux26' (without doing a ./configure or
any changes). Zaptel started compiling
Christopher L. Wade wrote:
Besides the obvious problems with using a remote network resource
as a timing device
Um, [hit head with idiot stick], I guess the incoming RTP stream would
be a 'remote network resource used as a timing device'. But in some
situations, we see the _problems_ with this,
We use NTP clock sources for a clock source on many of our physical T1
circuits. We use an outside stratum 1 clock source for our internal
server (stratum 2) and because we have our own server, we clock
everything else off of it (stratum 3).
Maybe I'm not familiar enough with the internals of
*Timing* is what is needed, not _time_. Two different things.
Besides the obvious problems with using a remote network resource as a
timing device, I don't think many NTP server admins would enjoy you
requesting a _time_ update on the order of 1000+ times a second? RTP
not relying on
On Tue, 12 Oct 2004, Remco Barende wrote:
On Tue, 12 Oct 2004, Christopher L. Wade wrote:
Remco Barende wrote:
OK, I tried compiling from tarball.
I got the zaptel-1.0.0.tar.gz. I untarred it. In that directory where the
tarball unpacked I just duid 'make linux26' (without doing a ./configure
or
Remco Barende wrote:
OK, thanks for explaining I thought it was something really serious :)
Yes, I did install the modules it created but it will not find the X100P
in the system
ztcfg -vvv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves:
Darren Sessions wrote:
We use NTP clock sources for a clock source on many of our physical T1
circuits. We use an outside stratum 1 clock source for our internal
server (stratum 2) and because we have our own server, we clock
everything else off of it (stratum 3).
Maybe I'm not familiar enough
Darren Sessions wrote:
Why not use an NTP timing source - go stratum 2 or 3. That should be
plenty for a stable clock source.
*Timing* is what is needed, not _time_. Two different things. Besides
the obvious problems with using a remote network resource as a timing
device, I don't
Write one :)
Altus Syman wrote:
So what do you suggest
Kanuri, Seshu (Company IT) wrote:
my bad, none of those applications really work for us and they are
too rudimentary of any value.
*From:* [EMAIL PROTECTED]
On Tue, 12 Oct 2004 08:57:52 -0500 (CDT), Joe Greco [EMAIL PROTECTED] wrote:
take on this is that with the X100P costing $100, and a Sipura 3000
costing $130 (all $USD), you'd likely need to have a Mac laying around
in order to justify this from a cost point of view, because even at a
loaded
Have your even had success sending couple pages at once without loosing
a part of the page?
Had the same problem with X100P and it's still unsolved.
Just wondering how I could synchronize timing with PSTN on the FXO card.
On Thu, 2004-10-07 at 19:41, Snezhana Bekova wrote:
Hi!
I have
Yes you are wrong. You seem to be combining two different methods of getting
SIP info out of a database. Pick 1. I use the perl script right now so here
is how to do that:
In order to use the perl script which can support 'ALL' sip abilities, use
this table:
CREATE TABLE sip_perl (
id
On Oct 10, 2004, at 10:08 AM, Wolf Paul wrote:
dean collins [EMAIL PROTECTED] wrote:
However, those of us not working with hefty corporate budgets may not
have the option of spending
$100 for a test machine when there's a more cost effective option
available.
I'd seriously suggest, in your
I want to dowload cvs of v1-0 with cvsup and was wondering what the options
file will look like to make this happen.
I am assuming the some thing on the line *default release=cvs tag=.
- options file for cvsup to download cvs head
*default host=cvs.digium.com
*default
Title: Message
G'Day
All,
Newbie here. How can
I go about troubleshooting a fast busy when I dial my the phone number on my *
server?
Thanks.
Ferg
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I've got a Linksys BEFSR41 at home with RoadRunner service. I'm pretty sure
it doesn't do QoS. I'm using WinXP Pro and not sure if it does QoS. I'm
using SJ Phone and...(follow the pattern).
I have to stop all network traffic on my machine if I want to have any hopes
of making a clear call. But I
For what it's worth, I just shutdown my PC based gateway last week and
replaced it with an Efficient Networks 5861 ADSL router. The 5861 is
billed as a Business Class DSL Router. It comes with Stateful
Firewall, DHCP, NAT, VPN, and QoS (WFQ), among other things. I have
not setup the QoS
Currently, if I briefly press the flash hook on my phone, the caller
is placed on hold. I would like for the channel to hangup if I do
this instead, never placing a caller on hold (I'll be using
call-parking instead). I disabled threewaycalling that is supposed to
control this, but it doesn't
sip users can register with *. Now, is it possible to relay the sip
signalling thru * the same way as sip proxies?
m. smadi
Benjamin on Asterisk Mailing Lists wrote:
On 08 Oct 2004 18:07:25 -0400, m. smadi [EMAIL PROTECTED] wrote:
i am used to using sip express router as sip proxy, but i
I think you mean SIP Gateway. You can forward the SIP off to a SIP Provider by
specifiying it in your sip.conf file as :
[mysipprovider-out]
type=peer
secret=password
username=2345
host=something.hcc.net
fromuser=2345
nat=no
then in your extensions.conf file:
i.e.
exten =
On Tue, 2004-10-12 at 17:01 +0900, Benjamin on Asterisk Mailing Lists
wrote:
On Mon, 11 Oct 2004 18:16:14 -0500, Steven Critchfield
[EMAIL PROTECTED] wrote:
2 cards is the highest number recommended.
That's not entirely correct.
It should be
2 cards is the higher number recommended
This will vary on the provider I think.
Telenor in Norway at first did not calculate PPPoE headers in the
purchased bandwidth. Later on, they changed their mind and increased the
speed because of the PPPoE headers.
The other followed this trend here in Norway, to keep up with the
competition.
Switching to DSL would require me to get a phone line, which kinda defeats
the purpose of doing VoIP. =)
Matthew
- Original Message -
From: Ryan Wilkins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, October 12, 2004 10:18
Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
when calculating bandwidth requirements on DSL, does the DSL
technology used (bridged,PPPoE,PPPoA,whatever) matter?
The techonology won't affect bandwidth as much as it affects latency
and jitter which are rather more important for VoIP.
Will
Steven Critchfield wrote:
Funny since there is only 4 real IRQ lines on a PCI bus. They are A, B,
C, and D. If you have more than 4 slots on a PCI bus, then you are most
definately reusing a real IRQ wire.
That is a drastic oversimplification; each PCI slot has only four IRQ
lines, but there is
[EMAIL PROTECTED] wrote:
Currently, if I briefly press the flash hook on my phone, the
caller
is placed on hold. I would like for the channel to hangup if I do
this instead, never placing a caller on hold (I'll be using
call-parking instead). I disabled threewaycalling that is supposed
to
On Tue, 12 Oct 2004, Steven Critchfield wrote:
Funny since there is only 4 real IRQ lines on a PCI bus. They are A, B,
C, and D. If you have more than 4 slots on a PCI bus, then you are most
definately reusing a real IRQ wire.
As for if PPC could handle it, I haven't seen any drivers.
Hi,
Has anyone had any luck getting one of the new ZyXEL
P2602HW routers working with *??
These units look good on paper: DSL modem, 802.11g, 4
Port Ethernet, 2 x ATA plus all the bells and whistles
in the firmware.
It has 2 different SIP clients built in and I was able
to get them registered
Not with an area served by Covad. Speakeasy uses Covad to deliver the
DSL service and Covad recently introduced their OneLink service which
does NOT require an active phone line for DSL services. What they do
is charge $6/mo over the cost of regular service and run a dry pair to
your
If you mean phone service rather than a phone line, then your
statement isn't correct.
SpeakEasy has a service they call OneLink which allows you to get DSL
without phone service. It's an additional $6/month over their normal
DSL rates.
- |Daryll
On Tue, 12 Oct 2004 10:38:46 -0500, Matthew
Subject pretty much says it all...
Thanks.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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Not necessarily - Speakeasy now offers naked DSL for a $10 or so premium.
Might want to check that out.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matthew
Boehm
Sent: Tuesday, October 12, 2004 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial
On 10/12/2004, Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED] wrote a lot of stuff.
Good god ben, don't you ever go to sleep? It must be, what, 2am in Japan
now?? Heheh.. i would have thought you'd be pretty pooped out by now,
what with the long threads and flame wars on the digium vs clone
On 10/12/2004, Ferguson, Michael [EMAIL PROTECTED] wrote:
G'Day All,
Newbie here. How can I go about troubleshooting a fast busy when I dial
my the phone number on my * server?
You might also want to check your hardware. What do you have running on
the box? More details would help us out in
1: FXS
2: FXS
3: none
4: none
--On Tuesday, October 12, 2004 07:18 -0400 Michael George
[EMAIL PROTECTED] wrote:
What is the configuration of the TDM400?
Port 1 -
Port 2 -
Port 3 -
Port 4 -
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Mine doesn't even Ping. Debian Woody 3.0 w/ a 2.4.25 kernel.
--On Tuesday, October 12, 2004 13:24 +0200 Altus Syman
[EMAIL PROTECTED] wrote:
Got the same problem on a linux redhat 7.3 box
But everything hangs
Its also runs mail
You can ping it but cant connect on any port,not eve the gui
On Tue, 2004-10-12 at 17:44 +0200, Peter Svensson wrote:
On Tue, 12 Oct 2004, Steven Critchfield wrote:
Funny since there is only 4 real IRQ lines on a PCI bus. They are A, B,
C, and D. If you have more than 4 slots on a PCI bus, then you are most
definately reusing a real IRQ wire.
Just as a what if...
Lets say I have a 250 phone rollout. I have three incoming T1 lines
(however thoes are usually setup) with say 1000 phone numbers available
to me. Every phone is currently analog, but I would like to move to a
VOIP based setup when the prices become comperable.
What am
Thanks. Resolved.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Flynn
Sent: Tuesday, October 12, 2004 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Fast Busy
On 10/12/2004, Ferguson, Michael [EMAIL
Hello,
look at
http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy
harry
--- Brian Wilkins [EMAIL PROTECTED] a écrit :
I think you mean SIP Gateway. You can forward the
SIP off to a SIP Provider by
specifiying it in your sip.conf file as :
[mysipprovider-out]
type=peer
Stan Brinkerhoff wrote:
Just as a what if...
Lets say I have a 250 phone rollout. I have three incoming T1 lines
(however thoes are usually setup) with say 1000 phone numbers
available to me. Every phone is currently analog, but I would like
to move to a VOIP based setup when the prices
You can purchase multi FXS channel banks. Check them out at ATACOMM.com
they sell them there. You will be able to get the 250 analog ports you
need from that.
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
On Tue, 2004-10-12 at 12:26 -0400, Stan Brinkerhoff wrote:
Just as a what if...
Lets say I have a 250 phone rollout. I have three incoming T1 lines
(however thoes are usually setup) with say 1000 phone numbers available
to me. Every phone is currently analog, but I would like to move to
Roadrunner as far as I know does not support QOS. Even if you had routers,
switches, etc that support qos, you r internet connection doesn't so it
would not do any good.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Tuesday, October
Hey
I get this error when I want to compile the cdr_mysql module:
mainserver asterisk-addons-1.0.1 # make
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`
cdr_addon_mysql.c:24:22: asterisk.h: No such file or directory
make -C format_mp3 all
make[1]: Entering directory
I have this issue if I start and stop * without totally rebooting. If i
reboot and leave * running then I have no problems.
- Original Message -
From: Michael Loftis [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, October
On Tue, 12 Oct 2004, Steven Critchfield wrote:
On Tue, 2004-10-12 at 17:44 +0200, Peter Svensson wrote:
This is not correct. Each pci slot has four physical interrupt lines, A-D.
The implementation is free to supply four separate interrupt lines to each
card, i.e. the interupt lines are
On 11-Oct-2004, Alex Barnes wrote:
I had/have exactly the same problem with my X100P / TDM400P dev setup.
I'm also having exactly the same problem with a TDM400P I received
yesterday. I'm starting to suspect that seeing it work after swapping
PCI slots is a placebo effect. Without moving the
I have Asterisk connected to a channel bank via a t100p card. There
excessive sidetone generated on the analog side due to an impedance mismatch
- I am very close to my serving CO which brings the line down to about
150ohms and the channel bank is expecting 600ohms. However, the very loud
hello Matthew,
I was wrong -:) but retrieving all sip info from
database would be better than running a perl script on
every Asterisk box in order to rebuild a
sip_additionnal.conf.(??)
so I have to create the table run the perl script in
order to create or overwrite a sip-additionnal.conf
but I
I believe retrieving in real-time is being worked on and should be done soon.
Developers are almost finished working on RealTime.
include = sip_additional.conf in [general]
On Tuesday 12 October 2004 05:26 pm, harry gaillac wrote:
hello Matthew,
I was wrong -:) but retrieving all sip info
Hi Folks,
I just try to get * 1.0.0 compiled on a SLACKWARE 10.0 box. * 0.9.1 did
compile and work without any problems. But now, I run into an compile error
which I just can't get resolved.
ZAPTEL compiles OK, LIBPRI complies OK, but then during compilation of
ASTERISK:
.
.
.
gcc -pipe
I have seen this behaviour as well with the t100p and tdm04b. I have
to power down, reboots don't work. For cards as pricey as these
you would think they would flush / refresh on a reboot.
ps. still trying to get the t100p to work in a data/voice
environment with little or no luck or
Well it's almost 100% sure that it's the zaptel drivers. Because I just
came back and the box's filesystem is hosed. Won't even boot now.
Hopefully I can get all my work back from /etc.
I'm really worried about putting Zaptel hardware into production now, if I
can't make a smoking gun point
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