On Wed, Mar 26, 2008 at 11:00:21PM -0500, Cody Jarrett wrote:
I have an old zapata tormenta 2 quad port pci card. I'd like to get it
working and play with it but was curious to see if that was possible.
Does anyone know if it will work for 2.6 kernels, or where I can find
decent drivers?
Hi,
I am having problem with my Asterix server. It does not hand up after play
the voicemail. The scenario is this: 1. I make a call to Asterisk's PSTN
number; 2. After recording, I hang up and make the same call again.
The first call would go through nicely with the voicemail recording, but
Hi All;
I need to buy one IAXy device, but I discovered that
it supports only g711 and ADPCM codec, so I was wonder
that it does not support g729 or GSM?!
Anyway, what is that ADPCM and how much it consumes
bandwitdh? Also, asterisk support such codec? What its
name in the configuration?
Any
Hi All,
I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17
and using it to make SIP calls.
I have a configuration of Asterisk which serves the users in a
particular domain, say internal.com
I would like to make a SIP call from [EMAIL PROTECTED] to
[EMAIL PROTECTED]
I have
Hi!
I'm a new member in VoIP world. I want to implement a VoIP PBX using
asterisk/tribox in the office but I have one doubt. Which is the best way to
use actual PSTN lines in the VoIP PBX? Using a box as Linksys SPA400 or
installing a PCI card (Rhino/Digium...) into the server? Which
Hi All;
I have been chocked just when I saw some posts talking
about how much the IAXy is bad :) -
So I would like to ask, did any one try it later and
wether it is good or not? I am asking this because I
need to use it as it is NAT Transparent (as I read
also, and I did not try it to see how
On Thu, Mar 27, 2008 at 3:58 AM, Juan Antonio Ibañez Santorum
[EMAIL PROTECTED] wrote:
Hi!
I'm a new member in VoIP world. I want to implement a VoIP PBX using
asterisk/tribox in the office but I have one doubt. Which is the best way to
use actual PSTN lines in the VoIP PBX? Using a box as
Most likely, you don't have any hangup detection available or
configured. If these are analogue lines, you will almost certainly need
to configure busy detection in order to figure out that the call has
been terminated.
Do some Googling for asterisk busy detection
mark morreny wrote:
Hi,
some better faxes handling lie 0 delay assured, important with hylafax for
example
2008/3/27, Juan Antonio Ibañez Santorum [EMAIL PROTECTED]:
but which would be the best option to include 4 PSTN lines into a VoIP
enviroment? As I can see that SPA400 (4 FXO Ports) is a cheaper option than
but which would be the best option to include 4 PSTN lines into a VoIP
enviroment? As I can see that SPA400 (4 FXO Ports) is a cheaper option than
a 4 ports PCI card (Rhino/Digium/Sangoma...). Would I get any benefit on
using a PCI card instead of a box as SPA400?
Regards
It depends on whether you want to deal with Zaptel or SIP/IAX trunks. My
personal choice would be the TDM400P but thats purely because I know it.
Barring any problems with IRQ sharing/conflicts, setup is fast and easy, and
it just makes sense to me to have all PBX-related hardware in one box
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote:
Do yourself a favor and upgrade a Asterisk 1.4 which has a proper
implementation of DTMF. It's likely your SIP provider upgraded to
something which does not recognize the DTMF tones from Asterisk 1.2.
I've upgraded to 1.4.18 (along with
Hi
I am not sure why this is happening or whether it has anything to do with my
iaxmodem setup. When receiving a fax via iaxmodem, I got an error message
saying *chan_iax2.c:7542 socket_process: Call rejected by 127.0.0.1: Busy*
From faxstat -s, I get:
JID Pri S Owner Number Pages Dials
I used to have hundred big HP 9000 boxes running HP-UX 11.0.
Having some open th entrailsls of those big boys and due surgery was a
damn good feeling. The also came with very very good HW diagnostics and
had some place you cold send KERNEL Dumps on troubled system that was
often a life saver
I had sent this to Digium Sales and cant get a response from them, I
don't. know what that means..
So I thought I would ask it hear since i know others have struggled with
this.
I an considering using *your High Density T*1 cards on a number of
servers we are considering purchasing. The
can anyone help me. I'm finding the softphone which can trigger web
browser and use callerid to go web page
You don't say on what OS you need it to run.
Mine is for Windows and support receiving URL (ex.:
Dial(IAX2/7003|20|trw|http://asterisk.org)
You can get it here :
Anyone have some up-to-date (within the past 3 months) on Asterisk and
the 7971. Searched voip-info, Google, etc., etc., to no avail.
Documentation I found was scattered, vague. Thanks in advance.
--
J. Oquendo
SGFA #579 (FW+VPN v4.1)
try doing a sip debug peer XXX (the problematic extension)
and then send a call to him till fail, then see the log, or send a piece
here.
On Thu, Mar 27, 2008 at 3:10 AM, Al lists [EMAIL PROTECTED] wrote:
Nope,
Coded is Ulaw on both sides and also this issue happens occasionally with
no
How do you get notifications ?
Is this thru one of the add on packages HP sells for the box ? Which One ?
Could you be more specific about what you mean by a recovery CD
and hod do you get console access below multi used to do recovery ??
What is integrated ILO BIOS Access sounds cool.
What O/S
Friday March 28th;: Astlinux
How to be a part of the conference via PSTN or SIP :
http://www.VoipUsersConference.org
Astlinux is a custom Linux distribution that is centered around the
goal of providing Asterisk. Astlinux is highly optimized for Asterisk
both in commercial and embedded systems.
Hi,
I am simulating the sending of fax using sendfax through voip to reach an
Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax
machine at ZAP/2. It seems like I am not able to establish any handshake
with the physical fax machine using the sendfax program. Does anyone
David Nedved wrote:
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote:
Do yourself a favor and upgrade a Asterisk 1.4 which has a proper
implementation of DTMF. It's likely your SIP provider upgraded to
something which does not recognize the DTMF tones from Asterisk 1.2.
I've upgraded
On Thu, Mar 27, 2008 at 6:48 AM, Al Baker [EMAIL PROTECTED] wrote:
I had sent this to Digium Sales and cant get a response from them, I
don't. know what that means..
So I thought I would ask it hear since i know others have struggled with
this.
I an considering using *your High
On Thu, Mar 27, 2008 at 8:16 AM, Darrick Hartman (lists)
[EMAIL PROTECTED] wrote:
David Nedved wrote:
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote:
Do yourself a favor and upgrade a Asterisk 1.4 which has a proper
implementation of DTMF. It's likely your SIP provider upgraded to
Notifications can be done either thru SNMP traps or SMTP. Insight
Manager is free from HP, but any SNMP trapper can work with alerts.
The recovery CD is just a build that reloads the majority of the system
with a static ip. We backup off site to one of our servers via FTP.
ILO access is an
On Thu, 2008-03-27 at 06:48 -0400, Al Baker wrote:
I an considering using *your High Density T*1 cards on a number of
servers we are considering purchasing.
Two-port T1 cards? Four-port T1 cards?
Could you please clarify *WHICH* of the above listed *PCI slots* are
suitable for use with
responses inline
bilal ghayyad wrote:
So I would like to ask, did any one try it later and
wether it is good or not? I am asking this because I
need to use it as it is NAT Transparent (as I read
also, and I did not try it to see how much it is
transparent).
Thousands and thousands and
I was trying to trap SIGHUP, but could be another signal because it didn't
work.
I'm doing this
class MyScript():
def logsignal(self,signum, frame):
self.putCDR()
def run(self):
signal.signal(signal.SIGHUP, self.logsignal)
def putCDR():
--- Darrick Hartman (lists) [EMAIL PROTECTED] wrote:
Try adding this line in the general section of extensions.conf
autofallthrough=no
The default behavior in 1.2 was no. In 1.4 it changed to yes. That
will be your simplest fix (without seeing your dialplan). Asterisk
is
moving on
On Wed, Mar 26, 2008 at 4:02 PM, Artifex Maximus [EMAIL PROTECTED] wrote:
Is it possible play background sounds while talking?
I would like to make an outgoing campaign with the possibility playing
sounds in background by command. But the extra is I would like to
choose which sound to be
bilal ghayyad wrote:
Hi All;
I have been chocked just when I saw some posts talking
about how much the IAXy is bad :) -
So I would like to ask, did any one try it later and
wether it is good or not? I am asking this because I
need to use it as it is NAT Transparent (as I read
also, and I
Thanks, got it working. Also, does the zapata tormenta 2 card have
only T1/E1 ports, or are they also FXS/FX0 ports?
Cody Jarrett
IT Freedom
direct 512.351.4965
[EMAIL PROTECTED]
office 512.351.7990 : fax 512.351.7991
On Mar 27, 2008, at 1:42 AM, Tzafrir Cohen wrote:
On Wed, Mar 26, 2008 at
On Thu, Mar 27, 2008 at 09:57:39AM -0500, Cody Jarrett wrote:
Thanks, got it working. Also, does the zapata tormenta 2 card have
only T1/E1 ports, or are they also FXS/FX0 ports?
No. Just E1/T1 ports. Analogs ports are different beasts.
--
Tzafrir Cohen
icq#16849755
--- Eric Wieling [EMAIL PROTECTED] wrote:
Inband only works with the ulaw and alaw codecs.
I think you might be onto something here. I don't have any explicit
allow or disallow lines, just taking the defaults. I've got plenty of
bandwidth and CPU, I'm much more concerned about calls going
On Thu, 27 Mar 2008 06:48:58 -0400, Al Baker [EMAIL PROTECTED] wrote:
I an considering using *your High Density T*1 cards on a number of
servers we are considering purchasing. The vendor lists that his system
has:
PCI Express*: two x8 slots*, t*wo x8 low profile slots*; *PCI-X:
On 27/03/2008, David Nedved [EMAIL PROTECTED] wrote:
So now it seems 1.4.18 is doing the same as 1.2.27 -- working for the
most part but completely ignoring DTMF on incoming SIP calls.
Perhaps you now need to delve deeper. Capture a UDP trace between your
VoIP provider and Asterisk, and
I've got two DL385s and a DL320, and they all rock. iLO especially rocks, but
to leverage the full functionality, you'll need to get the Advanced License,
which opens up full blown remote console capabilities (via Java). It's a
separate piece of hardware that, as long as the server PSUs have
This will be 4 T1s to a card.So, just so I am not confused.
In my original e-mail I said
I an considering using *your High Density T*1 cards on a number of
servers we are considering purchasing. The VENDOR lists that his SYSTEM has:
PCI Express*: two x8 slots*, twoo x8 low profile slots*;
On 08:02, Thu 27 Mar 08, Al Baker wrote:
How do you get notifications ?
Is this thru one of the add on packages HP sells for the box ? Which One ?
Could you be more specific about what you mean by a recovery CD
and hod do you get console access below multi used to do recovery ??
What is
No actually I gave it a LOT of thought and I and even asked two
different techs who repair PCs
and both said what the to vendors are saying is not sufficiently clear
that I would
make a purchasing decision based on what you have in hand from them
But, nice try at the cheap shot.
Steve Totaro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Al Baker
Sent: Thursday, March 27, 2008 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question about PCI Slots for
DIGIUMs Boards
No actually
I have a queue I configured as strict and a cron
script I use to QueueAdd and QueueRemove agents
according to my company's requirements. Usually I have
2 or 3 agents at a time and the ring strategy is
ringall.
These agents use non-open-source Windows softphones
that do not let you configure it so
Hi All,
For the most part, the PBX works as it should. Occasionally people complain
that they call and the PBX doesn't pick up. Other times it looks like the
call is answered by Asterisk but I still hear ringing and I start listening
to the IVR menu a few seconds into it.
As for Asterisk not
On Thu, Mar 27, 2008 at 01:40:48PM -0300, Gonzalo Servat wrote:
Hi All,
For the most part, the PBX works as it should. Occasionally people complain
that they call and the PBX doesn't pick up. Other times it looks like the
call is answered by Asterisk but I still hear ringing and I start
Sean Dennis wrote:
bilal ghayyad wrote:
Hi All;
I have been chocked just when I saw some posts talking
about how much the IAXy is bad :) -
So I would like to ask, did any one try it later and
wether it is good or not? I am asking this because I
need to use it as it is NAT Transparent
bilal ghayyad wrote:
Hi All;
I need to buy one IAXy device, but I discovered that
it supports only g711 and ADPCM codec, so I was wonder
that it does not support g729 or GSM?!
Anyway, what is that ADPCM and how much it consumes
bandwitdh? Also, asterisk support such codec? What its
name
You will never get latency on a network low enough for echo to be
perceived as sidetone (like on analog). If you want to get rid of echo
you must cancel echo.
Mojo with Horan Company, LLC wrote:
Sean Dennis wrote:
bilal ghayyad wrote:
Hi All;
I have been chocked just when I saw some
On Thu, Mar 27, 2008 at 1:56 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
Any suggestions??
I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2.
A freshly-built Asterisk? Built vs. zaptel 1.4.9.2 ?
Yes, I built 1.6.0-beta4 just recently with zaptel 1.4.9.2. As per your
suggestion on IRC,
I guess I've never run asterisk without ANY echo cans :) It's just that
the echo was minor enough that MG2 et. al did a fine job.
Thanks!
Moj
Eric Wieling wrote:
You will never get latency on a network low enough for echo to be
perceived as sidetone (like on analog). If you want to get
Ok - that is even more confusing , since one of them looks like it MIGHT
be a PCI slot
in the list PCI Express*: two x8 slots*, t*wo x8 low profile slots*;
*PCI-X: 64-bit/100MHz*
On Thu, Mar 27, 2008 at 08:32:28AM -0400, Steve Totaro wrote:
People on the list (mainly dev) want you to test, find bugs, jump
through hoops, and post to Mantis (where you bug might just be closed,
or a general feeling of You are wrong. All of this testing is free
of course due to the
On Thu, 2008-03-27 at 12:16 -0400, Al Baker wrote:
==
Here is the Confusion - 1) Can ANY of your QUAD cards go in the PCI
Express - two x8 slots - Yes or NO ?
Yes, the TE420B will fit. It's a 4-port T1/E1/J1/PRI card with echo
cancellation on-board. It only fits
Aadilkhan Maniyar wrote:
Hi All,
I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17
and using it to make SIP calls.
I have a configuration of Asterisk which serves the users in a
particular domain, say internal.com
I would like to make a SIP call from [EMAIL PROTECTED]
On Thu, Mar 27, 2008 at 2:38 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Thu, Mar 27, 2008 at 08:32:28AM -0400, Steve Totaro wrote:
People on the list (mainly dev) want you to test, find bugs, jump
through hoops, and post to Mantis (where you bug might just be closed,
or a general
Are you looking to purchase a server or just looking for a card for an
existing server?
Either way, post the specs from the manufacturer (and also make sure
those slots are open)
You will get your answer pretty quickly, and much less painfully this way.
Thanks,
Steve Totaro
On Thu, Mar 27,
I had a customer using an IAXY (old gen) for an FXO fax machine and it
worked almost all the time so it cannot be that bad.
Maybe because the fax was very old and did not have high transmit rates.
Thanks,
Steve Totaro
On Thu, Mar 27, 2008 at 2:11 PM, Mojo with Horan Company, LLC
[EMAIL
On Thu, Mar 27, 2008 at 02:58:31PM -0400, Steve Totaro wrote:
I am a user and a high level integrator, none of what you mention
applies to me. Maybe in a lab if I had time...
If you are a high-level integrator, then it seems to me you make direct
profit off the backs of the developers you
I have a cisco 7960 phone. Worked fine in the office.
I took it home. At home I have a linksys router that the phone is
plugged into.
The linksys router has DHCP enabled. I am getting the following error on
the console from the 7960.
I have tried it with nat=yes and nat=no in the sip.conf file.
We have a location that is having a really odd issue. We have a sangoma
POTs card. We are running software echo cancellation with the card
(through asterisk) to try to eliminate some major echoing problems. I've
turned on both EC and echotrain, which seemed to have gotten rid of the
echo for the
That's probably just someone at the NSA snooping your lines and playing tricks
on you...
g
--J
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rob Schall
Sent: Thursday, March 27, 2008 4:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I wanna say that's the echotraining taking effect.
What it does is try to cause some echo so it can dynamically reconfigure the
levels on the fly -- right at the start of the call. I know this happens
with digium cards -- not sure if the Sangoma cards behave the exact same
way. It's only at the
Hi,
I am investigating an issue with voicemail and realtime.
What we are seeing is the following:
1. Caller calls in and goes to an IVR
2. Presses 101 to go to voicemail
3. app_voicemail start and tries to connect to the database trhough
res_config_mysql. However, it takes too long to be able to
Enable NAT on the phone itself and leave it enabled in *.
Jerry Geis wrote:
I have a cisco 7960 phone. Worked fine in the office.
I took it home. At home I have a linksys router that the phone is
plugged into.
The linksys router has DHCP enabled. I am getting the following error on
the
them, then a few seconds lagging behind, you'll hear a muffled (darth
vader) version of the same thing.
I had a similar experience where people claimed it sounded like a
'Dalek' (yes, in UK). The sound is somewhat similar to Darth Vader, I
suppose.
This was down to a buggy
What are you trying to do? I run a 7970 here with SIP.
Thanks,
Matt
On Thu, Mar 27, 2008 at 7:02 AM, J. Oquendo [EMAIL PROTECTED] wrote:
Anyone have some up-to-date (within the past 3 months) on Asterisk and
the 7971. Searched voip-info, Google, etc., etc., to no avail.
Documentation I
Question: is anyone planning on going to the Cluecon convention this
year? (www.cluecon.com http://www.cluecon.com/ ) I'm hoping to go
this year and I'm hoping to meet other OSS telephony users and
developers. BTW, Anthony Minessale said that there is a need for
Asterisk speakers, so if you're
It's not bad in the sense of stability (well the original ones are
claimed to have overheating issues..).
But its that it lacks ANY features. The IAXy has no features at all.
Also no security, it MUST be placed behind a firewall, as the
configuration doesn't have any sort of security whatsoever.
On Thu, Mar 27, 2008 at 6:32 PM, Vieri [EMAIL PROTECTED] wrote:
I have a queue I configured as strict and a cron
script I use to QueueAdd and QueueRemove agents
according to my company's requirements. Usually I have
2 or 3 agents at a time and the ring strategy is
ringall.
These agents
calllimit in sip.conf and you are done
Vieri escribió:
I have a queue I configured as strict and a cron
script I use to QueueAdd and QueueRemove agents
according to my company's requirements. Usually I have
2 or 3 agents at a time and the ring strategy is
ringall.
These agents use
mark morreny wrote:
I am simulating the sending of fax using sendfax through voip
Ooops. Please see:
http://hylafax.sourceforge.net/docs/fax-over-voip.pdf
Thanks,
Lee.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Asterisk Users,
I am running Asterisk 1.4.11 on Debian
Etch system with the TDM03B wildcard. I recently purchased a
SPA-962 and SPA-932- the sidecar for our receptionist. After reading
many forum postings on how to configure the side car, I uprgraded the
SPA-962 software to 5.1.18(SC)
link with the same problem:
http://www.asteriskguru.com/archives/asterisk-users-tdm2400-hardware-echo-cancel-vt96394.html?highlight=tdm2400
nobody can solve the problem ?
Vu AnhTuan [EMAIL PROTECTED] wrote:
hi you,
I'm having problem with voice quality on my trixbox using
I have a question about DHCP and boot server supporting more than 1
model of Polycom phones.
According to Polycom standards, Polycom phone boots up to get a DHCP
address and at the same time getting a boot server string (with username
and password) to logon to boot server to download SIP, bootROM
For this, I would recommend using a smart DHCP device, which supports
the passing of 'option 66' - for example, the edgemarc series of
routers.
With that, you could pass ftp://user1:[EMAIL PROTECTED] via dhcp
in order to provision the phone, and different credentials if you are
concerned about
We have BLF buttons working fine on the SPA932 side-car. What does
show hints tell you under Asterisk, and what syntax did you use when
configuring the side-car buttons?
John Meksavan wrote:
Asterisk Users,
I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B
wildcard. I
All Polycom phones use the same firmware and bootroms - one reason why
the sip.ld is so damn large for them.
Lee, John (Sydney) wrote:
I have a question about DHCP and boot server supporting more than 1
model of Polycom phones.
According to Polycom standards, Polycom phone boots up to get a
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