I was following the instruction on
http://www.voip-info.org/wiki-Asterisk+non-root to re-install my
Asterisk as non-root when I had the following questions/issues:
1) Use your system's preferred method of adding a new user. Examples:
Red Hat: adduser -c Asterisk PBX -d /var/lib/asterisk -u
On Wed, May 14, 2008 at 09:14:44PM -0700, Roberto Milani wrote:
does the /tmp directory need to have some specific kind of mode/
ownership?
mine is linked to /private/tmp and is lrwxr-xr-x root admin
Yes, It is normally 1777 (all readable/writable/executable, but with the
sticy bit, which
Robert McNaught schrieb:
Does anyone know how to apply a style sheet to the polycom automatic
provisioning XML files?
Why should applying a stylesheet be different than for any other
XML files?
Even better, does anyone know of a web-based XML editor where you can
just edit the files from
On Thu, May 15, 2008 at 06:17:12PM +1000, Lee, John (Sydney) wrote:
I was following the instruction on
http://www.voip-info.org/wiki-Asterisk+non-root to re-install my
Asterisk as non-root when I had the following questions/issues:
For those wondering what the fuss is all about, look at:
He
On Wed, 14 May 2008 17:06:54 -0400, Alexander Lopez wrote:
SS7 helps carriers maximize the use of the circuits that interconnect
them with others. Instead of using a channel and having it open for 30
seconds as the call is setup, user gets signaling (busy, ringing, not in
service), and call
Lee, John (Sydney) schrieb:
I was following the instruction on
http://www.voip-info.org/wiki-Asterisk+non-root to re-install my
Asterisk as non-root when I had the following questions/issues:
1) Use your system's preferred method of adding a new user. Examples:
Red Hat: adduser -c
Hi All,
Whenever i try to start FOP using script
./op_panel_redhat.sh start given in directory /usr/local/op_panel-snapshot/init
I got the following error:
Starting Flash Operator Panel: execvp: No such file or directory
Lee, John (Sydney) wrote:
I was following the instruction on
http://www.voip-info.org/wiki-Asterisk+non-root to re-install my
Asterisk as non-root when I had the following questions/issues:
1) Use your system's preferred method of adding a new user. Examples:
Red Hat: adduser -c Asterisk
Hello ppl,
Are the channel names generated on 'Dial's supposed to be unique?
I see the channel names repeating on my asterisk box. I just wanted to confirm
this.
Can anyone point me to the lines of code where the channel name is
generated/calculated? I tried looking, but it looks like quite a
Benjamin Jacob wrote:
Are the channel names generated on 'Dial's supposed to be unique?
I see the channel names repeating on my asterisk box. I just wanted to
confirm this.
Can anyone point me to the lines of code where the channel name is
generated/calculated? I tried looking, but it
Benjamin Jacob schrieb:
Can anyone point me to the lines of code where the channel name is
generated/calculated? I tried looking, but it looks like quite a big maze.
ast_channel_alloc() in main/channel.c
---cut---
if (ast_strlen_zero(ast_config_AST_SYSTEM_NAME)) {
Philipp Kempgen schrieb:
Benjamin Jacob schrieb:
Can anyone point me to the lines of code where the channel name is
generated/calculated? I tried looking, but it looks like quite a big maze.
ast_channel_alloc() in main/channel.c
---cut---
if
Bryson Medlock wrote:
I'm trying to convince my employer to deploy an Asterisk based system,
but one member of the leadership team is against it. The rest of the
team is for it, but he's convinced them that we should find other
organisations in the Joplin, MO area who are using Asterisk
So I thought!! Thanks guys.
But a query with regards to this :
I need to send hangup commands based on these channel names only. So at any
given point of time, for 'n' ongoing calls, will these 'n' channel names be
different/ unique?
If not, using AMI, how do we hangup a given channel?
cheers
Benjamin Jacob wrote:
So I thought!! Thanks guys.
But a query with regards to this :
I need to send hangup commands based on these channel names only. So at any
given point of time, for 'n' ongoing calls, will these 'n' channel names be
different/ unique?
If not, using AMI, how do we
hi,
I have not tested that but I have seen 100 agents configure with asterisk.
thnks
Bhrugu mehta
On 5/15/08, gmail [EMAIL PROTECTED] wrote:
Is Asterisk practically stable and reliable for a larg Enterprise has say a
1 phones , is there any case study like this?
I have a Wildcard FXO: Wildcard X100P (clone) in my Asterisk (1.4.17)
machine and as of late, Caller-ID on it seems to be failing more
frequently than not. Sometimes I get callerid.c:613 callerid_feed:
Caller*ID failed checksum sometimes it fails without even that.
In Zapata.conf I have:
You'd probably want to run something else to handle your registrations like
OpenSER with that many phones.
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bhrugu Mehta
Sent: Thursday, May 15, 2008 8:31 AM
To: Asterisk Users Mailing List -
Brian J. Murrell wrote:
I have a Wildcard FXO: Wildcard X100P (clone) in my Asterisk (1.4.17)
machine and as of late, Caller-ID on it seems to be failing more
frequently than not. Sometimes I get callerid.c:613 callerid_feed:
Caller*ID failed checksum sometimes it fails without even that.
When I try to use ChanSpy, the following message is sent repeatedly to
the console (wrapped for readability):
WARNING[32125]: chan_sip.c:3709 sip_write: Asked to
transmit frame type 64, while native formats is 0x4 (ulaw)(4)
read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4)
This appears to
Matt Watson wrote:
You'd probably want to run something else to handle your registrations like
OpenSER with that many phones.
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bhrugu Mehta
Sent: Thursday, May 15, 2008 8:31 AM
To: Asterisk
On Thu, May 15, 2008 at 5:30 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, May 15, 2008 at 06:17:12PM +1000, Lee, John (Sydney) wrote:
5) Another article says that running as non-root will prevent ToS being
used. What is ToS? Do I need to be concerned?
Anybody wants to write something
Is there a way to force asterisk to ignore the first ring of a call without
using Wait() ?
When I active *72 call forward on my analog lines from the telco, they always
send a single ring and then do the forwarding. Asterisk picks up essentially a
dead line and rings the phones which gets
Alright guys and gals,
I'm a little lost, I'm primarily a SIP/IAX based guy, and have ended up
with a Zap installation. Everything was fine with our old provider when
we were using PRI, but the new provider screwed up on provisioning and
we've been temporarily stuck with a pair of EM Wink T's.
Hello
in extentions.conf I have the following menu
[voicemenu-custom-3]
comment = testmenunew
alias_exten = 6004
include = default
exten = s,1,Answer
exten = s,2,Background(thank-you-for-calling)
exten = s,3,Agi(agi://10.10.10.155/noaction)
exten = s,4,Hangup
There is a windows server who
Hello,
I have quite a bit of experience with EM Wink T1s, and I have seen
the problem you describe twice. In both cases it was either the
carrier's equipment or the wiring somewhere between the carrier shelf
and your equipment.
In one case it was water in the line that would seem to cause the
Matt Florell wrote:
Hello,
I have quite a bit of experience with EM Wink T1s, and I have seen
the problem you describe twice. In both cases it was either the
carrier's equipment or the wiring somewhere between the carrier shelf
and your equipment.
In one case it was water in the line that
On Thursday, May 15, 2008 11:11 AM - Sherwood McGowan said
...
we've been temporarily stuck with a pair of EM Wink T's. Ever since
then, we've been dropping 1-2% of all calls (in or out) and even more
strange, when a call gets dropped, a phantom call was being
generated on
the incoming
Is there any reason you don't want to use Wait()?
However, I would use WaitForRing() myself - its also a great solution on dirty
analog lines where you receive phantom calls.
That being said, I don't know how to do it without using some form of Wait.. as
far as I know zapata.conf doesn't
I hate to bring up an old thread, however, I'm implementing SLA as well.
I've got SLA working, my tunk executes slatrunk(line1), and my polycom 650
phone rings on the SIP subscribed line (button 1)
I'm assuming slatrunk sends the calls to the SIP/station1 SIP device, so
the call will always
Yes, perhaps a script would always be better than hand-touching these
files, and getting an XML editor only really makes it easier on the
eyes.
On the same subject, I have noticed that Snom and Linksys phones do
not support FTP provisioning - only TFTP and HTTP. With TFTP being an
insecure
On Thu, May 15, 2008 at 12:59 PM, Don Pobanz
[EMAIL PROTECTED] wrote:
On Thursday, May 15, 2008 11:11 AM - Sherwood McGowan said
...
we've been temporarily stuck with a pair of EM Wink T's. Ever since
then, we've been dropping 1-2% of all calls (in or out) and even more
strange, when a call
Matt Florell wrote:
Hello,
I have quite a bit of experience with EM Wink T1s, and I have seen
the problem you describe twice. In both cases it was either the
carrier's equipment or the wiring somewhere between the carrier shelf
and your equipment.
In one case it was water in the line that
I am confused how TFTP is less secure than HTTP. TFTP does not allow any
browsing, ever. Neither technologies will allow the device to
authenticate before downloading a configuration file, and both are
easily secured by only permitting connections from specific hosts.
Robert McNaught wrote:
Steve Totaro wrote:
On Thu, May 15, 2008 at 12:59 PM, Don Pobanz
[EMAIL PROTECTED] wrote:
On Thursday, May 15, 2008 11:11 AM - Sherwood McGowan said
...
we've been temporarily stuck with a pair of EM Wink T's. Ever since
then, we've been dropping 1-2% of all calls (in or out) and
Hi all,
What is maximum number of three party conferences can a quadcore 3GHz
system can handle? All the parties a setup with G.711 codec.
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asterisk-users mailing list
To
Wai Wu wrote:
Hi all,
What is maximum number of three party conferences can a quadcore 3GHz
system can handle? All the parties a setup with G.711 codec.
___
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asterisk-users
Hello,
The capacity greatly depends on the rate of calls entering and leaving
those conferences.
I see that you do call center systems so I would guess that the rate
would be fairly rapid.
It is really something you have to test and see. Using VICIDIAL in
performance testing mode I have gotten
In many call center applications, conferences are usually long and only
a small number them are necessary in any given time.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Florell
Sent: Thursday, May 15, 2008 2:24 PM
To: Asterisk Users Mailing List
did that and still no email
is asterisk logging something somewhere about errors in saving files
or so?
nothing shows up in the /tmp directory anyway
I have verbose and debug set to 100 in the CLI but I see no error
messages
HELP!
Roberto
On May 15, 2008, at 1:29 AM, Tzafrir Cohen wrote:
Everyone-
We're looking at using some Citel gateways to serve one of our sites (40
extensions, Toshiba phones). I've found that people seem to like the product
from demos, but I was wondering how many have some of the gateways in
production and if they seem to do the job for the long run.
Sherwood McGowan wrote:
Just a quick question, wanted to see if anyone knew where the
menuselect app stored your choices.
I think it's menuselect.makeopts but I'm not sure...just thought
someone might know.
Sherwood McGowan
P.S. I'll post here if I figure it out before there's a
Just a quick question, wanted to see if anyone knew where the menuselect
app stored your choices.
I think it's menuselect.makeopts but I'm not sure...just thought someone
might know.
Sherwood McGowan
P.S. I'll post here if I figure it out before there's a response :)
Whoa - you need some highly reliable, TELCO quality iron with some 1st
class support for that.
Do you realize what your downtime in that environment would would cost
you ?
Look, * is cool , fun an customizeable etc.
But it IS NOT carrier grade hardware and it is NOT software produced in
On Thu, May 15, 2008 at 10:08 PM, Robert McNaught
[EMAIL PROTECTED] wrote:
The way I understood it is that TFTP does not allow you to set a
username and password in a URL
like tftp://username:[EMAIL PROTECTED] is not possible
when setting option 66
Is it not possible to require a username
On Thu, 15 May 2008 10:23:14 -0700, Robert McNaught wrote:
Yes, perhaps a script would always be better than hand-touching these
files, and getting an XML editor only really makes it easier on the
eyes.
On the same subject, I have noticed that Snom and Linksys phones do
not support FTP
The items most people do not address are:
- QA - How do You tell if you you having Jitter,Packet Loss etc BEFORE
the user scream
- Disaster Recovery - from the small - DNS smokes - To Larger - * box
with 96 ports smokes
- Insuring EACH and EVERY piece ox network SUPPORT and USES QoS
-Vendor SLA
I will have a small shop with ~4 phones using an HP server with Asterisk on it,
it has two NICS and so I planned on plugging one into the cable modem, and the
other into the switch. I was going to let this box perform NAT for the company
but I am concerned about QOS for the VOIP portion.
I don't see why you couldn't use asterisk in a setup that large. It would
require a number of servers, and SER to handle the registrations, and call
routing and use asterisk for what its good at, ivr/vm.
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Limiting to HTTP would be OK if every customer had a static IP - if
you have small offices, then they maybe on DSL without static IP,
which makes that difficult - you could of course force your users to
have static IPs.
Robert
On Thu, May 15, 2008 at 1:45 PM, Atis Lezdins [EMAIL PROTECTED]
I have a lot of recordings from asterisk in a .gsm format. I would like
to play these files from a web browser (IE, firefox and opera)
What do I need to do in order to achieve this goal ?
Thanks
Julian
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A posting to the correct mailing list?
Or at least a post with the details of the issue? What OS? Can you
play these same .gsm files in any media player your OS might have?
On Thu, May 15, 2008 at 7:26 PM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I have a lot of recordings from asterisk in
On Thursday 15 May 2008 18:26:15 Julian Lyndon-Smith wrote:
I have a lot of recordings from asterisk in a .gsm format. I would like
to play these files from a web browser (IE, firefox and opera)
What do I need to do in order to achieve this goal ?
Allegedly, quicktime can play these files. I
Doesn't he mean something like when the recording happens, he'd like to go
http://192.168.1.1/recordings and then when he sees the list of *.gsm
recordings, he clicks on it, and the serverside starts playing it?
I think you'll need a Quicktime client (as as plugin to your brownser) on
your PC
Since, we're on the the topic of phones, and TFTPing.. if someone on this
thread has some knowledge of putting configs on Cisco IP Phone 7960, can
they please contact me off list?
I've done the configs via tftp, etc but anything into the speaker/handset
relating to voice doesn't work.
Mark Hamilton wrote:
Doesn't he mean something like when the recording happens, he'd like to go
http://192.168.1.1/recordings and then when he sees the list of *.gsm
recordings, he clicks on it, and the serverside starts playing it?
Yes, thanks ;)
I think you'll need a Quicktime client (as
No, no, no.
Don't try to play them directly as gsm files. Convert them to wav on
the fly, when demanded by the user from the webpage. Have a php, or
perl, or whatever script call sox, and push the wav to the user. sox
runs so fast that you can do the conversion on-demand. You can decide
what to
Hint:
core show file formats
If you specify your recording format as WAV you will get a gsm file
mangled to play in most browsers.
You can convert your existing files using SOX.
See http://www.callbandit.com/ for a sample gsm encoded WAV file.
___
--
On Thu, May 15, 2008 at 01:59:39PM -0500, Jonathan C. Bailey wrote:
We're looking at using some Citel gateways to serve one of our sites
(40 extensions, Toshiba phones). I've found that people seem to like
the product from demos, but I was wondering how many have some of the
gateways in
On Thu, 15 May 2008 15:39:26 -0600, Joseph L. Casale wrote:
I will have a small shop with ~4 phones using an HP server with Asterisk on
it, it has two NICS and so I planned on plugging one into the cable modem, and
the other into the switch. I was going to let this box perform NAT for the
Dear All
Please provide me the details how to configure hard phone with
Asterisk? If any one used Nortel phones
Or Cisco IP phone 7940 Please let me know how to configure at the
asterisk side as well as in the Device.
Thanks in advance
Regards
Ovia
Please do not print this email unless
What you're asking is well documented in many places.
Do your research.
Google is your friend.
www.voip-info.org also.
Michael
On Fri, 16 May 2008 08:19:30 +0530, [EMAIL PROTECTED]
wrote:
Dear All
Please provide me the details how to configure hard phone with
Asterisk? If any one
Hi Everyone, I'm pretty new to asterisk but coming from a call center
background; needless to say I am amazed. Here is my current dilemma; but
first some info on my setup. I have 3 public IP's from my provider...my
LAN sits under one behind a Sonicwall TZ-180, while my trixbox sits on
another
Hi all,
There is a setting called autopause in queue.conf to pause a queue
member if they fail to answer a call.
The autopause setting will pause the agent even when they are on the
line. I want to know if it is possible to pause the queue member only
when they don't answer after timeout?
ango
Hi Richard,
I'm not sure about the sonic wall issues, but for canadian providers try out
www.les.net
and
www.unlimitel.ca
We've had good success with both in the past.
Thanks,
Matt
On Thu, May 15, 2008 at 11:38 PM, Richard Spencer
[EMAIL PROTECTED] wrote:
Hi Everyone, I'm pretty new to
The docs as far as I can tell are not correct. E.g. Zaptel is required
(because it seems that it uses MeetMe) but none of that is documented.
So yes please do see if you can make the feature work and please post
a working example config for a Polycom phone.
On Fri, Nov 30, 2007 at 8:10 PM,
You SHOULD be concerned with QOS. All the way to an including the vendor
or your service cold really sucku
Michael Graves wrote:
On Thu, 15 May 2008 15:39:26 -0600, Joseph L. Casale wrote:
I will have a small shop with ~4 phones using an HP server with Asterisk on
it, it has two NICS
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