On 7/16/06, Erick Perez [EMAIL PROTECTED] wrote:
Hi,
I would like to know what kind of tests should I make in order to
document tone/configuration settings for analog cards and E1 cards
specifically for my country (Panama).
For example: Australia, Venezuela, etc, they have been documented and
Hi everyone,
I was trying to support SRTP in asterisk for our
Linksys IP Phones to prevent of ISP blocking issue.
I compiled successfully SRTP from
http://srtp.sourceforge.net/srtp.html
But i don't know from where i should start to
configure in Asterisk.
Could someone please give me the
Maxim Vexler [EMAIL PROTECTED] wrote:On 7/12/06, al gav <[EMAIL PROTECTED]>wrote: Hi all I need a help with asterisk+fax - fax to email I am trying to setup fax to email with asterisk with no success. I have asterisk 1.2.9.1 running on CentOS i have created extension 300 which should receive
Salve James, *!
On Sun, 16 Jul 2006, James Sturges wrote:
Don't know if it helps, but in AU you can tell the telco to place all calls
on 2 ISDN's at the same time.
That way you could have 2 ISDN lines on 2 ISDN cards (or Spans) and all
calls would be presented on both ISDN services.
I
On Jul 14, 2006, at 10:13 PM, Adrià Vidal wrote:
Someone using these phone Snom 300 with his own headset ?
We got horrible static noise on them?
P.D.
Got silence as answer from Snom by now... maybe on holidays or with
in the
European Football championship.
Have a look at this document:
Hi there,
Don't know if it helps, but in AU you can tell the telco to place all calls
on 2 ISDN's at the same time.
Same in Germany at Telekom: Standard BRI (2B+D) can be grouped together
onto the same number. But, I know just applications of this with the
point-to-point form of the
On Freitag, 14. Juli 2006 10:13 Adrià Vidal wrote:
Someone using these phone Snom 300 with his own headset ?
We used to but the quality was horrifying. Since we changed to Plantronics
Noise Cancelling headsets everything is wounderful.
We got horrible static noise on them?
Maybe the
Hi All
Has anyone got an annotated SEPmac.cnf.xml they are using successfully
with the 7970 (8.0.3 Sip) and Asterisk?
The SEPmac.cnf.xml files on the wiki are not annotated and although I've
managed to upgrade the phone firmware and get a partial registration better
info could speed it up.
Is
Could you possibly put up the relevant section(s) of your sip.conf? It sounds
like the DUNDi portion is set up properly, and obviously it's not going to find
an extension that doesn't exist.
Regards,
- Brad
From: [EMAIL PROTECTED] on behalf of Simon Woodhead
Abdul Lateef wrote:
Hi everyone,
I was trying to support SRTP in asterisk for our
Linksys IP Phones to prevent of ISP blocking issue.
I compiled successfully SRTP from
http://srtp.sourceforge.net/srtp.html
But i don't know from where i should start to
configure in Asterisk.
Could someone
http://news.yahoo.com/s/zd/20060714/tc_zd/183411
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Is there is a way to send Asterisk FLASH using DTMF? I am trying to
redial or dialing a new number without hangingup and start the whole
process again.
Thanks,
Osama
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Yes just use features.conf
On 7/16/06, Osama Kamal [EMAIL PROTECTED] wrote:
Is there is a way to send Asterisk FLASH using DTMF? I am trying to redial
or dialing a new number without hangingup and start the whole process again.
Thanks,
Osama
___
Wondering if someone else has ever done anything like this, or has any ideas
if it is in fact possible.
We currently record all our calls which are stored in gsm format. They are not
recorded by asterisk, rather a 3rd party system, but we would be looking at
using asterisk to implement this
Trixbox is only used as the client to simulate what we already saw
happening with a customer.
I don't think the fact that we used a Trixbox on the client side has
anything to do with the problem on the server side which is not using
Trixbox.
The on the server side Asterisk only sees the Trixbox
in my knowledge the only interaction with asterisk audio channels is
through eagi (refer to http://www.voip-info.org/wiki-Asterisk+AGI )
but as you can see there is no way to inject/add/mix audio
please tell me i'm wrong
On Mon, 2006-07-17 at 00:54 +0800, [EMAIL PROTECTED] wrote:
Wondering
Hello list
I'm trying to setup asterisk as an answering machine.
How can I set asterisk to Answer() incoming call ONLY after specified
count of ring cycles ?
In the current situation I have the PBX connected to a home line,
where POTS device are also connected on the same circuit. What I'm
Well, if the web interface copied the call to a standard name and you
had an extension using Playback or ControlPlayback to play that file
and then bridged the call -- maybe that wold work -- much of a kludge
though.
on Sunday 07/16/2006 mike([EMAIL PROTECTED]) wrote
in my knowledge the only
On Jul 16, 2006, at 11:36 AM, Maxim Vexler wrote:
Hello list
I'm trying to setup asterisk as an answering machine.
How can I set asterisk to Answer() incoming call ONLY after specified
count of ring cycles ?
In the current situation I have the PBX connected to a home line,
where POTS device
On 7/16/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jul 16, 2006, at 11:36 AM, Maxim Vexler wrote:
Hello list
I'm trying to setup asterisk as an answering machine.
How can I set asterisk to Answer() incoming call ONLY after specified
count of ring cycles ?
In the current situation I
Hi, I got this also and actually I still get this message. But i did realise the setup you ar trying to realise now. I wrote a little document on how to achieve this with trixbox. You can find it here:
http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdfNote that you should
On 16 Jul 2006, at 19:00, voiplist wrote:
Trixbox is only used as the client to simulate what we already saw
happening with a customer.
I don't think the fact that we used a Trixbox on the client side has
anything to do with the problem on the server side which is not using
Trixbox.
The on
Hi,
I need to monitoran asterisk server, so planning to use some tools which can initiate call to a number (for asterisk server)periodically and can interpret the response, is anything as such already available?? or any pointer??
thanks in advance
Nitin
Hi,
I'm setting up a new asterisk for an ecommerce company with cust sup dept.
The problem I'm having is with Roundrobin (and rrmemory also):
Let's suppose that I have 2 agents logged in into a queue. When a client
calls, and both agents are available. It rings the first one, but it
doesn't answer
Actually, at this point this info was more for the community as a
whole. We don't need to fix this now because the account code is right
and that's what is important.
On 7/16/06, Tim Panton [EMAIL PROTECTED] wrote:
On 16 Jul 2006, at 19:00, voiplist wrote:
Trixbox is only used as the
Stephen Murphy wrote:
My question is: How do I get the
current config files the phone is using off the phone?
AFAIK, you can't. :( You can only provide new configuration files from
your FTP/TFTP server. However, the Polycoms do strange things when
they've been configured in multiple
Hi all,
As I am starting to have stable live releases of a dialplan and
development work going on in parallel I need to have some sort of
regression test in place to ensure that no key functions of the current
dialplan are broken by a new version. Does anyone have pointers to the
best way to
So I can just install it over 1.2.9? This is what I did and everything seems
to be working fine.
Date: Sat, 15 Jul 2006 21:47:40 +1200
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.2.10 and
Is there any way to use the polycom phones to log into and out of queues? So
the polycom phone could show their current status in that queue? logged in /
logged out for example.
Thanks
Julian
Subject: RE: [asterisk-users] PRI dropouts
From:
Hi Santiago,
Unless it is a typo on the wiki, I think you want your queue.conf to be
like this:
member = Agent/@1
member = Agent/:2,1
That way you include group 1, and then include group 2 with
consideration of penalty. From the problem you are having it sounds like
the agent whose phone
Hi Julian,
If the 301's support ACD log in and log out, they should display a soft
button showing the current status of the phone, I know for sure the
601's do. Personally with our 601's I used two of the contact lines and
made my own log in and logout buttons and wrote my own script to log
Y'know, I was thinking about a similar idea recently, primarily because I do
a lot of work with dialplan based apps. It would be great if there was a way
to set up a _complete_ call (meaning it would include what digits to enter
when, etc) in a test and then run it against the dialplan being
Sharon Lim wrote:
Hi there,
I would like to ask, is it possible to group sip user? Means group A with
sip user 100,200 and group B with sip user 100,200?
thanks in advance.
in your dialplan, define the following variables:
GROUP_A=SIP/100SIP/200
GROUP_B=SIP/150SIP/200
and in your dial
Does Vicidial work together with Unicall/mfcr2 ?
Best Regards-- Bruno de Assumpção Loureiromsn: [EMAIL PROTECTED]
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In my experience PRI pass through setups have been false economy.
They seem to save a few dollars, but you still have to spend the money
to save it and they never run as well.
Paul Hales
--
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
bus: 03 8320 8100
mob: 0434 673 529
On
hmm...the group functions is to dial all the sip account, right. assuming if the dial plan is like exten = blah,1,Dial(${GROUP_A})exten = moreblah,1,Dial(${GROUP_B})then it will dial sip100 sip200 at the same time right? But i want to group it as different company. Is it possible?
Assuming, if 1
I don't know as I've never tested libunicall on any Asterisk system.
VICIDIAL will currently only work with Zap/SIP/IAX channels.
Can you install Unicall to use with USA T1s? Would it make sense to do
so for any practical purposes?
What does the Unicall channel show up as inside of asterisk?
I have a customer witha Polycom 501 phone behind a NAT. His phone is connected tohis Netgear router at home which in turn is connected to his cable modem. The phone is set up to register with our remote Asterisk server which is on a public, static IP address, with no NAT.
If we set qualify=yes,
On 7/17/06, Matt Florell [EMAIL PROTECTED] wrote:
I don't know as I've never tested libunicall on any Asterisk system.VICIDIAL will currently only work with Zap/SIP/IAX channels.Can you install Unicall to use with USA T1s?
No , it works with MFC/R2 signaling . There are some countries where ISDN
Yeah a bit messy I guess. I had been hoping for a simple solution, but
knew there most likely wasn't!
The one idea I did have would be to use some kind of SIP api on the web
backend. Then bring the backend extension into a conference, then from
the web api you would have to select the call to
According to your console output it looks like
there is some major latency. What is the average ping time from your
asterisk machine to the polycom phone?
- Original Message -
From:
Rana
Dutt
To: Asterisk Users
Sent: Sunday, July 16, 2006 6:51 PM
Subject:
At 06:25 PM 7/16/2006, you wrote:
exten = blah,1,Dial(${GROUP_A})
exten = moreblah,1,Dial(${GROUP_B})
then it will dial sip100 sip200 at the same time right? But i want
to group it as different company. Is it possible?
Assuming, if 1 have 2 company and want to have same sip account
context,
Hi people. I want to know about call forwarding. I
dial *72, and a message say me to dial the extension , I did, then the message
said is forward is UNCONDITIONLA . But when I call , it doesn't work the
forwarding.
Who can help me please.
Best Regards
Ever
Hi CF
I find that yes. The model of skipe was cracked.See link below:
http://politics.slashdot.org/politics/06/07/14/1514226.shtml
2006/7/16, C F [EMAIL PROTECTED]:
http://news.yahoo.com/s/zd/20060714/tc_zd/183411
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Hello,In some countries i found that they are blocking SIP port 5060so instead of this i change to another port 1221, and its workwell. But in one country the are not blocking SIP but they areplaying with RTP packets, if they filtered it is VoIP RTP theyare doing something called party cannot hear
On Jul 16, 2006, at 9:45 PM, Abdul wrote:
Hello,
In some countries i found that they are blocking SIP port 5060
so instead of this i change to another port 1221, and its work
well. But in one country the are not blocking SIP but they are
playing with RTP packets, if they filtered it is VoIP RTP
After several hours of searching the Internet, couldn't understand how can I integrate Asterisk with Sphinx voice recognition system. The sphinx software itself I've installed on my server.
I need help from those who have successfully done it and can guide me how to do it.
Thanks-- Zeeshan A
I am trying to install zaptel on dual Xeon processor but it gives error, saying 'You do not appear to have the kernel sources for your current kernel installed.make: *** [linux26] Error 1'
Googled for many hours, but nothing, except to use non smp kernel. How can I build zaptel for smp.-- Zeeshan
On Monday 17 July 2006 12:05 am, Zeeshan Zakaria wrote:
I am trying to install zaptel on dual Xeon processor but it gives error,
saying 'You do not appear to have the kernel sources for your current
kernel installed.
make: *** [linux26] Error 1'
Googled for many hours, but nothing, except to
You will have to install the kernel sources - what distro are you
running?
PaulH
On Mon, 2006-07-17 at 01:05 -0400, Zeeshan Zakaria wrote:
I am trying to install zaptel on dual Xeon processor but it gives
error, saying 'You do not appear to have the kernel sources for your
current kernel
How to install kernel sources?
On 7/17/06, Dennis Gilmore [EMAIL PROTECTED] wrote:
On Monday 17 July 2006 12:05 am, Zeeshan Zakaria wrote: I am trying to install zaptel on dual Xeon processor but it gives error,
saying 'You do not appear to have the kernel sources for your current kernel
On Mon, Jul 17, 2006 at 12:06:20AM -0500, Dennis Gilmore wrote:
On Monday 17 July 2006 12:05 am, Zeeshan Zakaria wrote:
I am trying to install zaptel on dual Xeon processor but it gives error,
saying 'You do not appear to have the kernel sources for your current
kernel installed.
make:
On Sun, Jul 16, 2006 at 11:31:24PM +0100, Nic Hughes wrote:
Hi all,
As I am starting to have stable live releases of a dialplan and
development work going on in parallel I need to have some sort of
regression test in place to ensure that no key functions of the current
dialplan are
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