Michael Cargile wrote:
Another reason I am sure that Digium has not released a DS3 TDM card is
the fact that asterisk currently cannot handle that many channels. I am
speaking from experience on this. We have build before a predictive
dialer with 16 PRIs. In order to do this and not have
On Mon, Apr 7, 2008 at 1:49 AM, Klaverstyn, David C
[EMAIL PROTECTED] wrote:
Thanks for that. I have the timeout set to 3000 ms and I have been pressing
the *1 within 500 ms so I don't think it is related to that. As I can do it
over SIP but not ZAP does not make much sense to me.
On Mon, Apr 7, 2008 at 2:01 AM, Alex Balashov [EMAIL PROTECTED] wrote:
Michael Cargile wrote:
Another reason I am sure that Digium has not released a DS3 TDM card is
the fact that asterisk currently cannot handle that many channels. I am
speaking from experience on this. We have build
Steve Totaro wrote:
A T3 MUXed into 28 T1 PRIs in one, or a few trunk groups inherently
has redundancy. If a box dies, the calls are dropped (unless you are
doing reinvite) and any call backs go right to the
Ts that are not in alarm.
True - and if you're simply using CT3 as an economical
hi, all
i am using zma800p card( from zapmicro).
i create small ivrs.
when i call on fxs channel calls lended and ivrs start on that channel.
but when i use callerid app. from asterisk , doesn't displayed any
callerid on asterisk.
any suggestion.
thanks in advance.
Bhrugu mehta
Dear Steve Doug;
Sorry I did not understand any thing from your reply.
---
Bogen Rulez
On 4/5/08, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi;
Anyone knows (tried) to use Page for analoge
phone(zaptel channel - fxs)? If yes, how?
Regards
Bilal
Steve Totaro
Hi there,
We're experiencing the following problem on a handful of boxes and I'm
wondering if anyone has any pointers for me.
First up, hardware configs on boxes are the same: Dell 860, Dual Core
Xeon, 1Gb RAM, 2x 160Gb SATA's in Software RAID 1.
Various Trixbox versions - 2.2.3 and 2.4.2 to
On Mon, 7 Apr 2008, James Williamson wrote:
Snap,
Well, after trying to buying a TDM400P and then getting persuaded to buy
a TDM410P
because they no longer sell the 400 model I'd say I'm not impressed. It
took three 2.6
kernel builds (zaptel 1.4 won't even build with the latest kernel
On Mon, Apr 07, 2008 at 06:11:02AM +0100, James Williamson wrote:
Snap,
Well, after trying to buying a TDM400P and then getting persuaded to buy
a TDM410P
because they no longer sell the 400 model I'd say I'm not impressed. It
took three 2.6
kernel builds (zaptel 1.4 won't even build
On Sun, 6 Apr 2008 17:22:58 -0400, Jay R. Ashworth [EMAIL PROTECTED]
wrote:
Yes, I've seen that, and most of its arguments are specious, at best.
They amount to I am too stupid to find a mail user agent with List
Reply, and too lazy to switch to it.
Are there any MUAs (other than Microsoft's
On Mon, Apr 07, 2008 at 10:37:42AM +0100, James Williamson wrote:
Tzafrir Cohen wrote:
On Mon, Apr 07, 2008 at 06:11:02AM +0100, James Williamson wrote:
Snap,
Well, after trying to buying a TDM400P and then getting persuaded to buy
a TDM410P
because they no longer sell the 400 model
Google is your friend. I discovered very quickly what they were talking
about by googling.
bilal ghayyad wrote:
Dear Steve Doug;
Sorry I did not understand any thing from your reply.
---
Bogen Rulez
On 4/5/08, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi;
Anyone
On Mon, Apr 7, 2008 at 6:36 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Apr 07, 2008 at 10:37:42AM +0100, James Williamson wrote:
Tzafrir Cohen wrote:
On Mon, Apr 07, 2008 at 06:11:02AM +0100, James Williamson wrote:
Snap,
Well, after trying to buying a TDM400P and then
Tzafrir Cohen wrote:
On Mon, Apr 07, 2008 at 06:11:02AM +0100, James Williamson wrote:
Snap,
Well, after trying to buying a TDM400P and then getting persuaded to buy
a TDM410P
because they no longer sell the 400 model I'd say I'm not impressed. It
took three 2.6
kernel builds (zaptel 1.4
On Mon, Apr 7, 2008 at 2:46 AM, Alex Balashov [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
A T3 MUXed into 28 T1 PRIs in one, or a few trunk groups inherently
has redundancy. If a box dies, the calls are dropped (unless you are
doing reinvite) and any call backs go right to the
Ts
Tilghman Lesher [EMAIL PROTECTED] writes:
And the arguments on the other side come down to I'm using an ISP
which can't correctly configure their mailserver, and I'm too lazy to set one
up myself.
How can the mail server fix a broken reply-to? It can remove it of
course, but that is rather
On Monday 07 April 2008 05:15:42 Rob Hillis wrote:
bilal ghayyad wrote:
Steve Totaro wrote:
On 4/5/08, bilal ghayyad [EMAIL PROTECTED] wrote:
Anyone knows (tried) to use Page for analoge
phone(zaptel channel - fxs)? If yes, how?
Bogen Rulez
Sorry I did not understand any
I have been working on a system where sometimes the users get a horrible
screeching noise that gets louder, typically as a call is hung up.
The system has 2 analog astribank channels with 8 FXO and 56 FXS ports, and
2 SIP phones, Asterisk 1.4.18, Zaptel 1.4.9.2. It is using the oslec echo
On Mon, Apr 7, 2008 at 8:09 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 07 April 2008 05:15:42 Rob Hillis wrote:
bilal ghayyad wrote:
Steve Totaro wrote:
On 4/5/08, bilal ghayyad [EMAIL PROTECTED] wrote:
Anyone knows (tried) to use Page for analoge
phone(zaptel
Hello,
Asterisk 1.4.19 crashes everytime using Realtime and SIP peers
gdb asterisk /tmp/coreXXX shows:
Program terminated with signal 11, Segmentation fault.
#0 0xb6148968 in find_peer (peer=0xb6042768 test, sin=0x0, realtime=1) at
chan_sip.c:2547
2547
On Mon, Apr 7, 2008 at 8:17 AM, Col Ferguson [EMAIL PROTECTED] wrote:
I have been working on a system where sometimes the users get a horrible
screeching noise that gets louder, typically as a call is hung up.
The system has 2 analog astribank channels with 8 FXO and 56 FXS ports, and
2
File a bug on Mantis.
On Mon, Apr 7, 2008 at 8:25 AM, Mindaugas Kezys [EMAIL PROTECTED] wrote:
Hello,
Asterisk 1.4.19 crashes everytime using Realtime and SIP peers
gdb asterisk /tmp/coreXXX shows:
Program terminated with signal 11, Segmentation fault.
#0 0xb6148968 in find_peer
Hi list,
sorry if this has been discussed in the past and is also posted twice to the
list,
but I couldn't find anything wise about it.
Since we had some trouble with the builtin hold function of some (all?) SNOM
320/360
phones, we decided to use the call parking feature in asterisk instead.
On Mon, Apr 07, 2008 at 10:17:33PM +1000, Col Ferguson wrote:
I have been working on a system where sometimes the users get a horrible
screeching noise that gets louder, typically as a call is hung up.
The system has 2 analog astribank channels with 8 FXO and 56 FXS ports, and
2 SIP phones,
I have a macro which I call when I want to ring the house phones. So
for example, when a Zap line rings it enters the dialplan in the
inbound-pots context which is as follows:
[inbound-pots]
exten = s,1,Set(CDR(accountcode)=...)
exten = s,n,Macro(check-incoming)
exten =
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 07, 2008 10:28 PM
Subject: Re: [asterisk-users] screeching noise on zap channels
On Mon, Apr 7, 2008 at 8:17 AM, Col
On Mon, Apr 7, 2008 at 8:36 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Apr 07, 2008 at 10:17:33PM +1000, Col Ferguson wrote:
I have been working on a system where sometimes the users get a horrible
screeching noise that gets louder, typically as a call is hung up.
The system
Thanks Tzafrir,
I will try that.
Cheers,
Col
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 07, 2008 10:36 PM
Subject: Re: [asterisk-users] screeching noise on zap channels
On Mon, Apr 07, 2008 at 10:17:33PM +1000,
On Mon, Apr 7, 2008 at 8:38 AM, Brian J. Murrell [EMAIL PROTECTED] wrote:
I have a macro which I call when I want to ring the house phones. So
for example, when a Zap line rings it enters the dialplan in the
inbound-pots context which is as follows:
[inbound-pots]
exten =
On Mon, Apr 7, 2008 at 8:14 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Mon, Apr 7, 2008 at 8:09 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Monday 07 April 2008 05:15:42 Rob Hillis wrote:
bilal ghayyad wrote:
Steve Totaro wrote:
On 4/5/08, bilal ghayyad [EMAIL
On Mon, Apr 07, 2008 at 08:44:50AM -0400, Steve Totaro wrote:
Was it a function of echo cancellation?
A case that was not handled well with echo cancellation in previous
versions.
http://svn.digium.com/view/zaptel?view=revrevision=4013
--
Tzafrir Cohen
icq#16849755
On Mon, 2008-04-07 at 08:48 -0400, Steve Totaro wrote:
Just ignore it. Exit on non-zero just means Hangup.
Strange. Hangup sounds like a perfectly valid and successful operation.
I wonder why make that a non-zero exit.
Maybe set
verbose 0 will make you feel better.
~sigh~ I want to see
Hi,
We have two PRI lines each with a huge span of numbers. We'd like to make
use of these lines/numbers for voice. In a typical setup, what would be the
perfect way to use this.
- Would one have a dedicated box that has the PRI cards configured running
Asterisk and it's only function is to
Steve Totaro wrote:
7 HP DL320s, RAID 1 with Quad Sangoma. Not a dozen but more than
twice that, 28 T1s. What is your cheaper solution? Also, have two
cold spares in the rack. DL 320s are cheap and rarely fail using
1.2.X. I actually cannot remember a single failure over years of
On Mon, Apr 07, 2008 at 02:46:48AM -0400, Alex Balashov wrote:
The point is that most people that want a DS3 interface really do want
to pump in a DS3's worth of calls, more or less, in which case they
really can't afford to have those DS1s going spare just for redundancy's
sake. And if
Hello,
I'm a little confused on DTMF.
A sip peer is registered on two Asterisk servers. No dtmfmode is set for
them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both
register on each other.
A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call
is
On Mon, Apr 07, 2008 at 10:48:34AM +0100, Horwich IT Services wrote:
On Sun, 6 Apr 2008 17:22:58 -0400, Jay R. Ashworth [EMAIL PROTECTED]
wrote:
Yes, I've seen that, and most of its arguments are specious, at best.
They amount to I am too stupid to find a mail user agent with List
Reply,
On Mon, 7 Apr 2008 11:35:43 -0400, Jay R. Ashworth [EMAIL PROTECTED]
wrote:
Question is: does Mailman *set* it?
Yes.
--
Godwin Stewart - Horwich IT services
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
On Monday 07 April 2008 07:05, Benny Amorsen wrote:
Tilghman Lesher [EMAIL PROTECTED] writes:
And the arguments on the other side come down to I'm using an ISP
which can't correctly configure their mailserver, and I'm too lazy to set
one up myself.
How can the mail server fix a broken
We have two PRI lines each with a huge span of numbers. We'd like to make use
of these lines/numbers for voice. In a typical setup, what would be the
perfect
way to use this.
You need to tell us for what purpose you're using these numbers? Are you
selling them out to the general public? Are
Hi Jean-Denis
2008/4/2, Jean-Denis Girard [EMAIL PROTECTED]:
Olivier a écrit :
Would you mind if I asked you this :
- Which card did you include in your home system ? Are you using an ISDN
BRI access ?
This is a basic BRI card with HFC chipset (Bewan Gazel 128)
Do you think it could
I have a server with 8 Xorcom Astribank, they're connected to the server via
a USB hub, my problem is: when a reboot the server the ID of the Channel
Bank changes, cousing a big mess on my server. Anybody know how can I solve
this ?
Best regards,
--
Guilherme Loch Góes
Visite nossa loja
On April 7, 2008 02:01:08 am Alex Balashov wrote:
A Lucent TNT Max outfitted with _plethoric_ VFCs might work okay. Apex
too, perhaps. Haven't tried to see how much it can handle when TDM-RTP
translation is required.
I'm curious; are the cpu/tdm/dsp requirements for 672 g729 rtp streams that
Andrew Kohlsmith (lists) wrote:
On April 7, 2008 02:01:08 am Alex Balashov wrote:
A Lucent TNT Max outfitted with _plethoric_ VFCs might work okay. Apex
too, perhaps. Haven't tried to see how much it can handle when TDM-RTP
translation is required.
I'm curious; are the cpu/tdm/dsp
On Mon, Apr 07, 2008 at 08:14:41AM -0400, Steve Totaro wrote:
(noted for future paging needs, I am doing a concrete factory next moth)
I hear *excellent* things about (and from :-) SoundSphere paging
speakers.
Cheers,
-- jra
--
Jay R. Ashworth Baylink
Chris,
Thank you for your detailed reply.
You need to tell us for what purpose you're using these numbers? Are you
selling them out to the general public? Are you using them in-house? If
you're selling access to them, are you selling predominantly to individuals
(i.e. single SIP devices) or
On Mon, Apr 07, 2008 at 11:11:34AM -0500, Tilghman Lesher wrote:
So the question comes down to, do you reply to the list more often or
do you reply off the list more often? Because the more frequent case
wins.
That *really* is not the question.
The question is: which causes more grief to
On Mon, Apr 7, 2008 at 10:46 AM, Alex Balashov
[EMAIL PROTECTED] wrote:
Steve Totaro wrote:
7 HP DL320s, RAID 1 with Quad Sangoma. Not a dozen but more than
twice that, 28 T1s. What is your cheaper solution? Also, have two
cold spares in the rack. DL 320s are cheap and rarely fail
On Mon, Apr 07, 2008 at 06:29:09PM +0200, Olivier wrote:
Hi Jean-Denis
2008/4/2, Jean-Denis Girard [EMAIL PROTECTED]:
Olivier a écrit :
Would you mind if I asked you this :
- Which card did you include in your home system ? Are you using an ISDN
BRI access ?
This is a
On Mon, Apr 07, 2008 at 02:12:03PM -0300, Guilherme Loch Waltrick Góes wrote:
I have a server with 8 Xorcom Astribank, they're connected to the server via
a USB hub, my problem is: when a reboot the server the ID of the Channel
Bank changes, cousing a big mess on my server. Anybody know how can
Hi!
If the caller hungs up while an eagi script is running, I can´t regiter the
cdr manually at the end of the script.
I tryied to trap SIGHUP but it didn´t work.
I want to register my own cdr into the script because I have a lot of data
that I need to put in the cdr.
The 'h' option or DeadAgi
Yeah, Asterisk I think would be more than capable of doing that. It'll
need some work to glue it all together. A lot of this would be written
as an AGI script, and PHP or so for the webpage part of it.
Sounds fun!
blackwater dev wrote:
We currently have an application used by the trucking
Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
I don't know about cordless, but for corded, I've had great success
with Plantronics H91N's.
- Noah
___
-- Bandwidth and
hi:
i'm a new of asterisk voip server, i compiling without problem asterisk 1.4.18,
and other software and component.
i create two extension 2 and 20100... and 3 voicemailMain
but i can't call any extension this is the logs
/var/logs/asterisk/messages
[Apr 7 13:25:19] WARNING[24402]
On Monday 07 April 2008 12:48, Jay R. Ashworth wrote:
On Mon, Apr 07, 2008 at 11:11:34AM -0500, Tilghman Lesher wrote:
So the question comes down to, do you reply to the list more often or
do you reply off the list more often? Because the more frequent case
wins.
That *really* is not the
On Mon, Apr 7, 2008 at 2:07 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Apr 07, 2008 at 02:12:03PM -0300, Guilherme Loch Waltrick Góes wrote:
I have a server with 8 Xorcom Astribank, they're connected to the server
via
a USB hub, my problem is: when a reboot the server the ID of
On Mon, 7 Apr 2008, Yanier Salazar Sanchez wrote:
[Apr 7 13:25:19] WARNING[24402] app_dial.c: Unable to create channel of
type 'SIP' (cause 3 - No route to destination)
Whatever device was the destination could not be found.
[Apr 7 13:26:51] WARNING[24408] chan_iax2.c: Unable to open IAX
Ruben Zamora wrote:
Hi,
I have a same problem, last week i was working with TE120 with a little
echo in some call, I replace the card
with a TE122B ( Included Echo Cancelation VPMADT032) and there was no
more echo in my call.
But know i have de same probelm with my incoming audio stream
i'm install libpri and zaptel
cd libpri
./configure
make make install
and zaptel too
cd zaptel
./configure
make make install
but when i execute modprob ztdummy
the system it's block
- Original Message -
From: Steve Edwards [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On Mon, Apr 07, 2008 at 03:43:30PM -0400, Steve Totaro wrote:
Maybe you could contact the guys from the app_rpt, they were having
the same issue with multiple USB URIs but came up with a solution.
Maybe it is the same solution you have or maybe something a little
simple.
This does well. We
i'm using debian etch r3
- Original Message -
From: Yanier Salazar Sanchez [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, April 07, 2008 4:35 PM
Subject: Re: [asterisk-users] new instalation os asterisk
i'm
try sangoma.com card, easy to use, hardware echo cancel, and carrier
quality.
matt
- Original Message -
From: James Williamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, April 06, 2008 10:11 PM
Subject:
Have u tried elastix.org
On Sun, Apr 6, 2008 at 9:02 AM, Thermal Wetland [EMAIL PROTECTED]
wrote:
On Fri, Apr 4, 2008 at 4:42 PM, Jonn R Taylor [EMAIL PROTECTED]
wrote:
I made some install scripts based on centos 4 or 5 like trixbox but
without all the junk. It does have some fax setup
Steve Totaro wrote:
Nobody said anything about unused DS1s except you. Please re-read the thread.
Perhaps I misunderstood?
What you appeared to imply in regards to the failover benefits of having
T1s broken out of an M13 mux into multiple gateway machines is that if
one of the gateways go
Hi,
Tzafrir Cohen a écrit :
On Mon, Apr 07, 2008 at 06:29:09PM +0200, Olivier wrote:
Do you think it could work with a Bewan Gazel 128 USB ?
I could get a hand on one and before diving into it, I would be very curious
to get your opinion on it.
I read carefully the thread you pointed me to
When I record calls they turn out really quiet. I can barely hear the
person on the other end. But when making the call, I can hear the
person on the other end just fine. I've played with txgain/rxgain and
that doesn't seem to help. TIA for help.
I would recommend the Plantronics CS70N
On Mon, Apr 7, 2008 at 11:47 AM, Noah Miller [EMAIL PROTECTED] wrote:
Any suggestion for a headset (cord and cordless) for IP601?
Any good (and economical) ones from Polycom or Platronics?
I don't know about cordless, but for corded, I've had great
Andrew Kohlsmith (lists) wrote:
On April 7, 2008 02:01:08 am Alex Balashov wrote:
A Lucent TNT Max outfitted with _plethoric_ VFCs might work okay. Apex
too, perhaps. Haven't tried to see how much it can handle when TDM-RTP
translation is required.
I'm curious; are the
Ruben,
Contact support at digium they have a release on a firmware that fixes
this and other issues with the VPMADT032.
Apparently it comes on newer zaptel drivers.
Good luck with your install.
Lex
On Mon, Apr 7, 2008 at 3:11 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Ruben Zamora
Hi All,
I thought I read a post a while back of a system call or something in
the dialplan whereby a call count can be incremented and spit out to a
text file.
Not like a group count of active channels.
What I would like to accomplish is have an incremental count of a
specific dialplan routine
Doug Lytle wrote:
Edwin Lam wrote:
in FaxDispatch:
FILETYPE=pdf
case $CALLID4 in
1000)
[EMAIL PROTECTED]
1001)
[EMAIL PROTECTED]
*)
[EMAIL PROTECTED]
esac
This is also incomplete,
actually there's no problems with the above script. it's just
$CALLID4
Lex
Thanks a lot. These morning i call Digium Support. One issue that i
miss in my before e-mail is that i have
my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my
MFC/R2.
Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall.
They told me they can help me because they dont have
Steve Totaro wrote:
Sorry for all the replies, I found the Digium PDF on Data mode.
http://www.modulo.ro/Modulo/docs/TE405-410P-user-manual.pdf
Good luck getting them to support it though ;)
I will post my Sangoma results tomorrow.
Thanks,
Steve Totaro
On Sun, Apr 6, 2008 at 10:49 AM,
When I downloaded the sip and bootrom from Polycom website, I noticed a
file called SoundPointIPWelcome.wav. However, I have no idea where and
when it was used. I played the wav file but I have never heard the
phone using this wav file before. Does anyone know what it is used for?
On Tue, Apr 8, 2008 at 12:06 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
When I downloaded the sip and bootrom from Polycom website, I noticed a
file called SoundPointIPWelcome.wav. However, I have no idea where and
when it was used. I played the wav file but I have never heard the
It's played at the completion of the boot process. It's always been
very quiet on the models I've worked with.
Thanks Erik. I can probably replace it with my beloved Mozart Symphony
no 40 :-)
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