On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote:
Hello,
On most SIP phones a conference call is done on the phone and is limited to 3
participants. Polycom phones has a configuration option to use a conference
server instead of the internal conferencing feature. I guess I need some
Kai-Uwe Jensen wrote:
There's also an app_swift available at http://www.loopfree.net/app_swift/
Thanks to all that responded. I've used app_swift as mentioned above and
it suits my needs.
Thanks again
Julian
___
--Bandwidth and Colocation
On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote:
Hello,
On most SIP phones a conference call is done on the phone and is limited to
3
participants. Polycom phones has a configuration option to use a conference
server instead of the internal conferencing feature. I guess I need
Use Snom phones.
We have had around 6 participants, without problems. In theory you should be
able to have around 12 people on a conference on a snom phone.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yehavi Bourvine
+972-8-9489444
Sent: 19.
Use Snom phones.
We have had around 6 participants, without problems. In theory you should be
able to have around 12 people on a conference on a snom phone.
I have a few Snom phones here - people do not like them...
Thanks, __Yehavi:
Peder @ NetworkOblivion wrote:
Group pickup / call pickup is the feature you want.You put everybody
in a group and if you want to grab a ringing phone, you just hit the
group pickup code.
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
Does this work with 1.4.1
Jon Schøpzinsky wrote:
Use Snom phones.
We have had around 6 participants, without problems. In theory you should be
able to have around 12 people on a conference on a snom phone.
I don't think this is true. The Snoms do not have enough
CPU power for 12 people in a conference *on the
With 6 people it works, we have tried it. The 12 people is, as I said, only in
theory, because, as you said, the CPU is probably not powerful enough.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: 19. marts 2007 09:57
To:
Hi everybody,
i've a E1 connection with 30 phone numbers, i'm using freepbx
(trixbox) with TE110P card,
when i call my sell phone just the first number of E1 is always showed
evenif i set up different trunk with different zapchannel
(g0,1,2,3...)
for example when i type 1MyPhonenumber , i get the
Hi Everyone,
Google IS my friend. I found the solution via Google on the second
glance ;-)
It seems that the USB latency was too high and you had to increase a
CAPI-Buffersize in chan_capi.h:
#define CAPI_MAX_B3_BLOCK_SIZE 500
(German instructions:
Hi,
after upgrading my server (Debian 3.1 + tdm400p + monoBRI) from
Asterisk1.2.9.1 to new Asterisk 1.4 (same hw, same os) I got the
voicemail sounds randomly chopped. I checked the sounds with a player
and they seem good. I made tests with default modules.conf settings
(noload =
cat5 cables are ok if you use straight cables.
crossed cables are different as ethernet signals use other pin layout than e1.
and beside the 'official' e1 crossed, there seems to be other layouts.
this has been discussed here, so browse the archives.
(my pc gives me headaches now, otherwise i
Hi list,
I want to set up special contexts for every sip user. But a context=XYZ
does not help in the perr definition as I have to provide a context in
the general section of sip.conf. This is my sip.conf:
[general]
port=5060
bindaddr=192.168.0.75
disallow=all
allow=ulaw
allow=alaw
Yuan LIU wrote:
Just noticed that no matter what the line condition is, zttool always
reports OK, so it's pretty useless. (In contrast, I'd get Red alert
if I unplug the line connecting to an X100P.)
This is the normal behavior. Only X100P will report the real status.
2007/3/17, Gordon Henderson [EMAIL PROTECTED]:
How about a mini tower type unit? I've just bought one of these:
http://www.asus.com/products.aspx?l1=9l2=40l3=121model=1017modelmenu=2
Or if you really need to put it in a rack, a fanless 1GHz Via processor in
a 1U rack fitted with a 2.5 laptop
Yehavi Bourvine +972-8-9489444 wrote:
Why not use the MeetMe feature of asterisk?
I need the person who initiated the conference call to call the others and join
them by herself. If I understand correctly, with the MeetMe you have to
initialize the conference and then people dial by
Yehavi,
Can you make a script that uses call files to get everyone into the
conference?
--
Warm Regards,
Lee
Possible, but looks too much cumbersome... However, that's a nice idea.
Thanks! __Yehavi:
Or, you can just transfer the calls into the conference room.
On 3/19/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Yehavi Bourvine +972-8-9489444 wrote:
Why not use the MeetMe feature of asterisk?
I need the person who initiated the conference call to call the others
and join
them by herself.
On 19 Mar 2007, at 12:29, Olivier wrote:
2007/3/17, Gordon Henderson [EMAIL PROTECTED]:
How about a mini tower type unit? I've just bought one of these:
http://www.asus.com/products.aspx?
l1=9l2=40l3=121model=1017modelmenu=2
Or if you really need to put it in a rack, a fanless 1GHz Via
Tim Panton wrote:
I really like the Mac Minis as small office servers, quiet, cool,
real UNIX, asterisk works on them.
The only downside is that you can't add PSTN cards.
Aren't there USB adapters available?
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied -
On 19 Mar 2007, at 14:31, Philipp Kempgen wrote:
Tim Panton wrote:
I really like the Mac Minis as small office servers, quiet, cool,
real UNIX, asterisk works on them.
The only downside is that you can't add PSTN cards.
Aren't there USB adapters available?
Anything needing a kernel
asterisk-users@lists.digium.comAsterisk 1.4
I have strategy= leastrecent and autofill = yes
I have 2 agents, one is answering a call and the other is free and have some
calls waiting in the queue.
Only when the first agent hangup the second agent receive the first call in
the queue.
It happends
New install, using TDM400p. wctdm is loaded, asterisk loads. Zaptel and
zapata.conf are from a working system, same model board, same module locations.
CLI command zap show status shows all OK, zap show channels shows nothing
defined.
Incoming calls show nothing on CLI, analog handsets have
On 3/19/07, equis software [EMAIL PROTECTED] wrote:
Asterisk 1.4
I have strategy= leastrecent and autofill = yes
I have 2 agents, one is answering a call and the other is free and have
some calls waiting in the queue.
Only when the first agent hangup the second agent receive the first call
Please send me any news about this or the bug number.
Thanks for your time.
On 3/19/07, BJ Weschke [EMAIL PROTECTED] wrote:
On 3/19/07, equis software [EMAIL PROTECTED] wrote:
Asterisk 1.4
I have strategy= leastrecent and autofill = yes
I have 2 agents, one is answering a call and the
HI,
I dont understand the syntax of the dial application when used like this:
Dial(Local/[EMAIL PROTECTED])
i want to know what is this Local doing instead of Tech like SIP, IAX,
H323?
--
Regards
Rizwan Hisham
Software Engineer
___
--Bandwidth and
Rizwan Hisham wrote:
I dont understand the syntax of the dial application when used like this:
Dial(Local/[EMAIL PROTECTED])
i want to know what is this Local doing instead of Tech like SIP, IAX,
H323?
SIP/200 would dial a device (the SIP user 200) whereas
Local/200 dials the extension
Hi joe,
seems like your card is working but not your modules...have you checked
if the led on the TDM400P are on?
Try to change the FXO/FXS modules, maybe they are not working good.
Giorgio
joe acquisto wrote:
New install, using TDM400p. wctdm is loaded, asterisk loads. Zaptel and
Yuan LIU wrote:
Just noticed that no matter what the line condition is, zttool always
reports OK, so it's pretty useless. (In contrast, I'd get Red alert
if I unplug the line connecting to an X100P.)
I'm using zaptel 1.2.15 on Linux 2.6.15-28 (also tested on 2.6.10).
Correct. The TDM400P
Thanks.
Turns out etc/asterisk/zapata.conf had two entries for [channels]. First one
was blank. I
had cut and pasted and somehow did not notice this utill I was on the phone
with Digium.
I guess asterisk reads the first and ignores the next one. Removing the first
one fixed the problem.
Does anyone have any idea on how to force cepstral to convert a number
to speech ?
I have noticed that sometimes it speaks the number correctly, and at
others it doesn't.
1) 787 is pronounced 7-8-7
2) 123 is pronounced one-hundred and twenty-three.
1) is wrong for what i need, 2) is
Oh man - the second I send this, I find the answer.
say-as type=currency12345.44/say-as
Sorry for the waste of bandwidth.
Julian
Julian Lyndon-Smith wrote:
Does anyone have any idea on how to force cepstral to convert a number
to speech ?
I have noticed that sometimes it speaks the number
Julian Lyndon-Smith wrote:
Does anyone have any idea on how to force cepstral to convert a number
to speech ?
I have noticed that sometimes it speaks the number correctly, and at
others it doesn't.
1) 787 is pronounced 7-8-7
2) 123 is pronounced one-hundred and twenty-three.
1) is wrong
From: Christophorus Laube [EMAIL PROTECTED]
Date: Mon, 19 Mar 2007 13:23:34 +0100
Hi list,
I want to set up special contexts for every sip user. But a context=XYZ
does not help in the perr definition as I have to provide a context in the
general section of sip.conf. This is my sip.conf:
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Date: Mon, 19 Mar 2007 11:26:56 -0500
Yuan LIU wrote:
Just noticed that no matter what the line condition is, zttool always
reports OK, so it's pretty useless. (In contrast, I'd get Red alert if
I unplug the line connecting to an X100P.)
I'm
Hi all,
I setup two * boxes with two sip phones (one to each * box). I can make
calls from one sip phone to the other via IAX both ways. However, the
dtmf tones are just oneway. I use rfc2833 for dtmfmode in the sip.conf
on both * boxes, and gsm for IAX. What do I need to achieve twoway dtmf
Hi everybody,
after installing hylafax iaxmodem i get this email
==
The HylaFAX software thinks that there is a problem with the modem
on device /dev/ttyIAX that needs attention; repeated attempts to
initialize the modem have failed.
Consult the server trace logs for more
younss azzayani wrote:
Hi everybody,
after installing hylafax iaxmodem i get this email
==
The HylaFAX software thinks that there is a problem with the modem
on device /dev/ttyIAX that needs attention; repeated attempts to
initialize the modem have failed.
This would be
Hi.
I am in the Brazil.
I have a problem on Unicall for Asterisk 1.4.
I am working with E1 with 30 channels.
Message when it receives calls:
chan_unicall.c:2603 handle_uc_event:unicall/1 protocol error.cause 32772.
My files:
/etc/zaptel.conf:
span=1,1,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
I am geet this error, I assume because I have zero digium hardware
installed. This is to be an entirely web based PBX.
Can anyone point me to an easy 123 for installing zaptel in dummy form?
I need music on hold for a VPS server.
Brad
-Original Message-
From: [EMAIL PROTECTED]
Question was off topic for the thread, but I'm feeling helpful today.
More of a 1234...
make install
modprobe usb-uhci
modprobe zaptel
modprobe ztdummy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brad
Sumrall
Sent: Monday, March 19, 2007 13:17
To:
On Mon, 19 Mar 2007, Olivier wrote:
2007/3/17, Gordon Henderson [EMAIL PROTECTED]:
How about a mini tower type unit? I've just bought one of these:
http://www.asus.com/products.aspx?l1=9l2=40l3=121model=1017modelmenu=2
Or if you really need to put it in a rack, a fanless 1GHz Via processor
On Mon, 19 Mar 2007, Brad Sumrall wrote:
I am geet this error, I assume because I have zero digium hardware
installed. This is to be an entirely web based PBX.
Can anyone point me to an easy 123 for installing zaptel in dummy form?
I need music on hold for a VPS server.
You have'd said what
Also forgot, ztdummy is not used with hold music, it would be used for
mixing audio in the meetme app.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darryl
Dunkin
Sent: Monday, March 19, 2007 12:41
To: Asterisk Users Mailing List - Non-Commercial
Julian Lyndon-Smith wrote:
Does anyone have any idea on how to force cepstral to convert a number
to speech ?
I have noticed that sometimes it speaks the number correctly, and at
others it doesn't.
1) 787 is pronounced 7-8-7
2) 123 is pronounced one-hundred and twenty-three.
You could
Hello,
I am wondering if someone can tell what components and modules I need to
effectively offer virtual pbx hosting using asterisk and wholesale
termination. I would appreciate all input.
Best,
Apostol
___
--Bandwidth and Colocation provided
We have a Teliax IAX trunk that we use as an overflow for our four
regular business lines into our local Asterisk PBX (Trixbox). We have
our Teliax account set up so that it goes to a Teliax voicemail box if
it cannot reach our Asterisk server, and we have the channel set up for
5 simultaneous
Is there any way to use Festival from script called by the ExternalIVR()
dialplan function?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided
Hi List,
Has anyone been successfull with getting H.323 to work with G729 ?
Thanks.
Dovid___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
On 3/19/07, Scott Plante [EMAIL PROTECTED] wrote:
work better in general. Is it the general experience on the list that
SIP is more mature and reliable than IAX? We like the fact that we don't
have to open inbound ranges of ports for IAX to work. We are in Atlanta
I've switched to using SIP on
Hi,
This is a tool that allows you at any time and any place of your Dialplan
or Dialout Call file to dial a specific extension at a specific context,
even if you are not currently in the specific context.
example:
you are at [from-internal] context and you can say:
[from-internal]
exten=
Hello List -
Here's the setup:
Mediatrix 1102 ATA (t38enabled) -- Asterisk 1.4.1 -- IP -- SIP GW --
TDM
The T38 call comes up perfect - I see the initial invite, followed by G711,
Re-Invite, T38 establishes, Fax Completes, T38 Stops, Call Down.
here's the problem - I see the following in my
I recently got Festival performing Text to Speech on my Asterisk system.
It is working great when I call from extension to extension in the
house. But when I dial in on my phone number (which comes in on a sip
registration to a Sonus server), I can not hear any sound. The
asterisk box thinks it
Olivier wrote:
I'm really after 1U-2U silent servers as I've got the feeling most of
them are too noisy for offices and most of our clients don't have server
rooms.
Try this:
http://www.tomshardware.com/2006/01/09/strip_out_the_fans/
-s
___
Scott Plante wrote:
We have a Teliax IAX trunk that we use as an overflow for our four
regular business lines into our local Asterisk PBX (Trixbox). We have
our Teliax account set up so that it goes to a Teliax voicemail box if
it cannot reach our Asterisk server, and we have the channel set
younss azzayani wrote:
Hi everybody,
after installing hylafax iaxmodem i get this email
==
The HylaFAX software thinks that there is a problem with the modem
on device /dev/ttyIAX that needs attention; repeated attempts to
initialize the modem have failed.
Consult the
http://www.crn.com.au/story.aspx?CIID=76033eid=4edate=20070320
The company developed Response Point to work alongside traditional phone
systems or voice-over-IP systems.
Continuing its recent foray into the market for digital communications
products, Microsoft on Monday introduced its first
Chris Carey wrote:
I recently got Festival performing Text to Speech on my Asterisk system.
It is working great when I call from extension to extension in the
house. But when I dial in on my phone number (which comes in on a sip
registration to a Sonus server), I can not hear any sound. The
Gordon Henderson wrote:
On Mon, 19 Mar 2007, Brad Sumrall wrote:
I am geet this error, I assume because I have zero digium hardware
installed. This is to be an entirely web based PBX.
Can anyone point me to an easy 123 for installing zaptel in dummy form?
I need music on hold for a VPS
Hi,
I have an strange problem and did not understand what happened here.
Iam using an SIP provider to call an analog POTS phone.
I orginate the call to the analog phone and send the call in an queue.
If the analog phone hangs up, the SIP provider did not send BYE and Asterisk
think the line
Stephen Bosch wrote:
Olivier wrote:
I'm really after 1U-2U silent servers as I've got the feeling most of
them are too noisy for offices and most of our clients don't have server
rooms.
Try this:
http://www.tomshardware.com/2006/01/09/strip_out_the_fans/
-s
The fans are in
Wow. Does that mean that someone calling into the system will be able
to use the Embedded voice recognition technology and halt the system
by saying stop now? Or will it just do that without anyone saying
it?
On 3/19/07, Dean Collins [EMAIL PROTECTED] wrote:
Is that FUD really necessary?
On 3/19/07, C F [EMAIL PROTECTED] wrote:
Wow. Does that mean that someone calling into the system will be able
to use the Embedded voice recognition technology and halt the system
by saying stop now? Or will it just do that without anyone saying
it?
On 3/19/07,
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, 19 March 2007 8:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Microsoft launches first PABX
Wow. Does that mean
I think yes, why you disagree?
On 3/19/07, mitcheloc [EMAIL PROTECTED] wrote:
Is that FUD really necessary?
On 3/19/07, C F [EMAIL PROTECTED] wrote:
Wow. Does that mean that someone calling into the system will be able
to use the Embedded voice recognition technology and halt the system
by
The Microsoft press release and links to photos etc.:
http://www.prnewswire.com/news/index_mail.shtml?ACCT=104STORY=/www/story/03-19-2007/0004548826EDATE=
--
George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102
www.netvoice.ca www.ip-centrex.ca
www.digium.ca
Steve Totaro wrote:
Stephen Bosch wrote:
Olivier wrote:
I'm really after 1U-2U silent servers as I've got the feeling most
of them are too noisy for offices and most of our clients don't
have server rooms.
Try this:
http://www.tomshardware.com/2006/01/09/strip_out_the_fans/
-s
The
Like someone else said, go for either really big (4U) or also really
small (fanless ITX). 4U rackmount with a fanless PSU, 120mm fans can
flow a decent amount of air and be nearly silent, you just need to
shop around a bit. A fanless ITX with a compactflash card is great for
a small
There are two ways that you can implement reliable faxing:
1) Implement end-to-end QoS and insure that latency is very low and
there is no jitter. Should be 95% reliable, more or less depending on
the quality of the link. Most people don't have the ability. It's not
possible with ADSL, not
SIP or IAX? What are the relevant configs?
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I think the best we can hope for is a stable release of OpenPBX. T38
gateway there is not a single report of it working, this is the holy
grail of T38 faxing. I don't know if perhaps it is only designed for
T38 to Zap only? Txfax and rxfax won't even compile with the newer
asterisk 1.2 and newest
C F wrote:
I think yes, why you disagree?
Has Microsoft actually ever come with such useful features?
It would be great to demonstrate the complete instability/insecurity of
Windows based servers by have it shut down automatically in front the
boss with a recorded message :D. Even better
Hi Yehavi,
Yes, this can be done. We are currently using this features. The Secretaries
making the calls to who ever her Boss wants to join the conference she then
just transfer the calls into the conference room. You can even annouce the name
of the newly arrived calls in the conference.
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