On 04/29/06 10:06 Josué Conti said the following:
is that if the agent transfers the call, for another user and this user
takes care of the call, the status of the agent in the show agents is
of that it the same continues speaking (talking to zap) with circuit
how are you performing the
Is there a way to monitor the DTMF tones on a channel?
I have a prepaid application working in asterisk. When the user dials a call and
wants to cancel the call before it is answered, there is now way to do it
without hanging up and redialling the access number.
Is there way to monitor a
[EMAIL PROTECTED] wrote:
I've been playing around with a new system I'm going to install in
another office. In setting up the Polycom's, I accidently used a new
power supply from a new 601 (24VDC) with an 600. The 600 only require
12VDC. Now, I get nothing on the screen of the 600 when I
Hi all,
Please excuse my newbie status
I need help in configuring a
mediatrix 1204 PSTN gateway with asterisk.
Basically each FXO port is configured with a SIP username and automatic
transfer extension, which should transfer incoming calls to an asterisk
extension. I created extensions
Hi,one of my WiFI phone has problem with the notify asterisk signal to me the following Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '192.168.100.124' does not implement 'NOTIFY'
In theory the phone support this function.Any idea?
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Frank Attard wrote:
I am pasting 3 SIP messages between the Mediatrix (192.168.0.27) and
Asterisk (192.168.0.6) upon an incoming call. Asterisk is returning 407
error.
407 is not an error. SIP errors are in the 5xx and 6xx range. 407 means
Asterisk is expecting the SIP device to provide
Benoit Panizzon wrote:
Hi
After migrating from 1.2.4 to 1.2.5 I noticed that:
show application dial
does not show the 'R' option anymore. Has this become an undocumented feature
I'm running 1.2.7.1 and I do show the 'r' option. I would suggestion
you remove the /usr/lib/asterisk/modules
Dinesh the agents they receive a call and this call will have to be transferred, them uses only functions hold and trnsf in device Polycom IP 301, below mine features.conf This problem, only occurs with calls that if they originate in the pilot of queue and when an agent receives and transfers. It
Il Neofita wrote:
Hi,
one of my WiFI phone has problem with the notify asterisk signal to me
the following
Apr 29 06:49:16 WARNING[6455] chan_sip.c: Host '*MailScanner warning:
numerical links are often malicious:* 192.168.100.124
http://192.168.100.124' does not implement 'NOTIFY'
In
Dear all,
Do anyone know to setup asterisk's SIP channel to use an outbound proxy
outside of asterisk's network to proxy the SIP message?
Thanks
Ray
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To
Raymond Chen wrote:
Do anyone know to setup asterisk's SIP channel to use an outbound proxy
outside of asterisk's network to proxy the SIP message?
This is documented in the sample sip.conf file in the configs directory
of your Asterisk source tree.
On Fri, 28 Apr 2006, Klaus Darilion wrote:
Back to ISDN BRI crossover cable. After reading some ISDN specs I came to the
conclusion a crossover cable should be:
3---4
4---3
5---6
6---5
Yes.
But I also found other pin layouts (e.g.
Indeed... I just don't understand Nufone... we provide VoIP
services... and have a contract inplace with our CLEC, we also have
backup sources for numbers, LD termination, etc. No backup plan =
BAD!
On 4/28/06, Kerry Garrison [EMAIL PROTECTED] wrote:
AMEN!!
Any consultant that DOESNT take
Well that's what I did and they seem to be operating just fine. The
CLEC told me even though they are the same CLEC, it is a different
switch.. but yeah hehe in theory I guess the timing would have to be
the same since THEIR switches are linked, eh?
On 4/28/06, [EMAIL PROTECTED] [EMAIL
Benoit Panizzon wrote:
On Friday 28 April 2006 15:32, Eric ManxPower Wieling wrote:
What does the R option do?
Indicate 'Ringing' as soon as the called party indicates 'Ringing'.
The 'r' option indicates 'Ringing' as soon as the connection is built, even if
the called party is not yet
I just read:
Certain options to the Dial() statement require that Asterisk is in the
media path, and consequently Asterisk will not let go of it: /t/, ''T,
h, H, w, W or L (with multiple arguments). Probably there are
more.
I had in my memory that r, R, m would also prevent a reinvite. Can
Hi,
I'm interested in anybody that is providing a phone support service
using an Asterisk system, with built in charging system.
I run a PC support company and use Asterisk at the home/office. I
would like to be able to provide technical support to my customers
using asterisk. However I want to
Mimmus wrote:
Hi,
I have a lot of GXP-2000 phones not registering with Asterisk server.
After two days of attempts, it seems that problem is due to the fact that
phones and server are not on the sme network.
Do you know if this is known issue?
We have GPX-2000s connecting via different
Hello all,
I have two test beds running the exact same version of asterisk 1.2.7.1,
latest of zaptel, libpri, etc..
Test bed #1 (Solaris 9,sparc ultra 5):
This one is closer to a production machine, in that it is connected to a
sip provider thru an iax2 connection and have an incoming DID
T.S wrote:
Hello all,
I have two test beds running the exact same version of asterisk 1.2.7.1,
latest of zaptel, libpri, etc..
Test bed #1 (Solaris 9,sparc ultra 5):
This one is closer to a production machine, in that it is connected to a
sip provider thru an iax2 connection and have an
T.S wrote:
Hello all,
I have two test beds running the exact same version of asterisk 1.2.7.1,
latest of zaptel, libpri, etc..
Test bed #1 (Solaris 9,sparc ultra 5):
This one is closer to a production machine, in that it is connected to a
sip provider thru an iax2 connection and have an
The client's needs are the mother of invention. We
have a client that currently uses a Cisco Call Manager and one of the features
they love was the Locate-Me function (or follow-me, or find-me, whatever you
want to call it) which basically rings their desk phone a few times then plays a
I agree with you, but I would like to find a way to use the notification. I tough that there was a work around.On 4/29/06, tom
[EMAIL PROTECTED] wrote:Il Neofita wrote: Hi, one of my WiFI phone has problem with the notify asterisk signal to me
the following Apr 29 06:49:16 WARNING[6455]
Mike Dent wrote:
My idea was for them to phone or login to a website and create a
support account. They can then top this account up with X amount of
credits, lets say 1 credit= 5 mins of support. Their account has a PIN
associated with it.
This is a typical prepaid system at work.
When
Hi everyone,
I've been trying to chase down a bug in Asterisk 1.2. I have 2
completely different setups which exhibit the same problem, and I'm not
sure what it is.
For instance, one machine is setup as a voicemail server. If you call
it, it says password..you put it in, and it says, You
Hi, I will try that thanks.Andrew
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I bought one of these. It's a great device. So good that I gave it to my
boss to use with Skype. It's far better than the speakerphone in the
Alcatel phone on his desk. We've used it with Skype and Gizmo.
Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
The following occurs during make asterisk-addons.
I'm ok with asterisk but debugging things like this isnt my strong point.
Can anyone give me a pointer?
Thanks
Dan Journo
[EMAIL PROTECTED] src]# cd asterisk-addons[EMAIL PROTECTED] asterisk-addons]# makemake -C format_mp3 allmake[1]: Entering
Hi,all.
Have read a lots of documents and wiki and topic there.. I get a solution for Large asterisk...
1,in IAX or SIP config file...set..
[general]
regcontext = iaxregistrations
[peer]
name=peer
regexten= 10001
2,in extensions.conf.
[default]
exten = _X,1,Macro(dundi-priv,${EXTEN})
Hi all, I was just
wondering ifanyone knows of any gotchas with respect to upgrading Asterisk
to the latest 1.2.7 ?
Is the procedure the
same? Config files remain intact? Just untar/make
install?
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
I have searched google and came up with too many options and packages
that may or may not work for my needs, most articles seem to be for
setting up routers. Maybe someone on the list can give me some better
insight.
I have monitoring turned on my shift eight (tm) (Asterisk ;-)) box for
all
Make clean, make make install. Just dont do make samples.
Dave Morrow wrote:
Hi all, I was just wondering if anyone knows of any gotchas with
respect to upgrading Asterisk to the latest 1.2.7 ?
Is the procedure the same? Config files remain intact? Just
untar/make install?
David
It's a little crude but you can
1: Use VLAN(ing) on the Cisco Switch to segment the traffic on an
addition 'LAN'.
2: Low Budget, Add a NIC on a separate network with the NAS.
3: Give me a bit, It'll come to me! :-)
SNIP!!
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I would not write a contract with a company that had these types of
issues http://voxilla.com/name-News-article-sid-166.html
Who eats $450,000?
Matt wrote:
Indeed... I just don't understand Nufone... we provide VoIP
services... and have a contract inplace with our CLEC, we also have
backup
Alexander Lopez wrote:
It's a little crude but you can
1: Use VLAN(ing) on the Cisco Switch to segment the traffic on an
addition 'LAN'.
The VLAN option would not work I dont think because the data is all
going out the same interface whether or not it has a VLAN tag
2: Low Budget, Add a
upgrading from what version?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
MorrowSent: Saturday, April 29, 2006 6:11 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] RE: Install/Upgrade
Hi all, I was
At 06:17 PM 4/29/2006, you wrote:
My question is, how can I throttle the FTP (Standard with dist)
transfers using out of the box CentOS4.3 (or any easy to use, low
learning curve package)? I thought about FTPing the files at less
frequent intervals but that just makes the issue less frequent
Ira wrote:
At 06:17 PM 4/29/2006, you wrote:
My question is, how can I throttle the FTP (Standard with dist)
transfers using out of the box CentOS4.3 (or any easy to use, low
learning curve package)? I thought about FTPing the files at less
frequent intervals but that just makes the issue
I upgraded my server from Fedora Core 4 to Fedora Core 5.
I was wondering if anybody else has run into the problem and know's the
fix?
I recompiled asterisk and if I don't have
the /usr/lib/asterisk/modules/codec_g729a.so
file in place it works.
I use or used to use the licensed G729 Codec from
What version of Asterisk are you using? If it's trunk then you'll have to
wait for the G729 codecs to be rebuilt with the new loader changes.
On 4/29/06 11:49 PM, Jason A. Kates [EMAIL PROTECTED] wrote:
I upgraded my server from Fedora Core 4 to Fedora Core 5.
I was wondering if anybody else
On Sat, 2006-04-29 at 21:17 -0400, Steve Totaro wrote:
My question is, how can I throttle the FTP (Standard with dist)
transfers using out of the box CentOS4.3 (or any easy to use, low
learning curve package)? I thought about FTPing the files at less
frequent intervals but that just makes
I know someone will suggest this should go on the -biz list,
but this is a one time event and not a business for me.
I have a new 2621XM router with
(1) NM-HDV
(1) VWIC-2MFT-T1
(4) PVDM-12 DSP modules
(1) ADSL WIC
(1) WIC-1DSU-T1
It was purchased for a prvate project that never got off the
This is the version reported on startup:
Asterisk 1.2.7.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
This is the list of packages I downloaded and compiled:
asterisk-1.2.7.1.tar.gz
asterisk-addons-1.2.2.tar.gz
asterisk-sounds-1.2.1.tar.gz
libpri-1.2.2.tar.gz
zaptel-1.2.5.tar.gz
hi, all,,,
there have something need to correct
1,in IAX or SIP config file...set..
[general]
regcontext = iaxregistrations
[peer]
name=peer
regexten= 10001
2,in extensions.conf.
[default]
exten = _X,1,Macro(dundi-priv,${EXTEN})
exten = _X,2,Playback(invalid)
exten =
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the time.
Of course his voice quality is like a morse code with dashes or dots of
connection time.
The next minute he calls me via Skype and it
I have some DIDs from NuFone (tollfree).
How can I switch them and to which provider? What is the cost for that?
What is the procedure for that?
bye
Ronald Wiplinger
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One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the time.
Of course his voice quality is like a morse code with dashes or dots of
connection time.
The next minute he calls me via Skype and it
On Sun, 2006-04-30 at 11:53 +0800, Ronald Wiplinger wrote:
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the time.
Of course his voice quality is like a morse code with dashes or dots of
Steve Totaro wrote:
I have searched google and came up with too many options and packages
that may or may not work for my needs, most articles seem to be for
setting up routers. Maybe someone on the list can give me some better
insight.
I have monitoring turned on my shift eight (tm)
Steve Totaro wrote:
I have searched google and came up with too many options and packages
that may or may not work for my needs, most articles seem to be for
setting up routers. Maybe someone on the list can give me some better
insight.
I have monitoring turned on my shift eight (tm)
Ronald Wiplinger wrote:
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the time.
Of course his voice quality is like a morse code with dashes or dots of
connection time.
If that's what is
Hi. recently I have been trying to setup a PRI on asterisk. Inbound
calls are working just fine but I am not able to make outbound
calls. Does anyone know what I need to change to make outbound
calls work? Right now the PRI is instantly hanging up on the outbound calls.
I have included
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