Hi
Where do i put this, and what shall i change do make it work for me?
/ralf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion
Sent: den 20 januari 2009 18:17
To: Asterisk Users Mailing List -
Hi,
I have several Aastra 57i phones connected to 1.4.22 version of asterisk and
when I call the queue these phones are part of I get few of these phones
ringing with a delay ... as much as 18 secs or not at all ...
What could be the problem?
Thanks
[Jan 22 03:06:19] VERBOSE[13842] logger.c:
Hi,
I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using
chan_ooh323 from asterisk-addons.
I am able to make a call from SIP Phone - Asterisk - Avaya - Station
(phone) and vice versa.
I am also able to make a call from SIP Phone - Asterisk - Avaya - PSTN.
However I face
To: All Snap users and future ADA users
I hope you enjoy the product under Digium's leadership. During my time
working on Snap I learned a lot about the Asterisk community, building
software products, and meeting customer needs. From this experience, I
took Snap and made it as lightweight,
Is there the possibility to increase the debug of an AJAM command?
If DTMF works on channel, and my command is queued successfully, what
can be the problem?
Thanks
On Thu, Jan 15, 2009 at 4:34 PM, nik600 nik...@gmail.com wrote:
Hi to all
i'm using PlayDTMF with AJAM, after the
Hi asterisk users,
I am in need of information about how to configure the
sip.conf and extension.conf for subscribers to support the dialog event
package rfc 4235. I am using asterisk 1.6.0.1 version.
The below are the configuration of sip.conf and extension.conf files
which I
try a answer() before the dial(sip/xxx)
and if you are using originate try local/ and start whit and answer()
2009/1/22 Steven J. Douglas stev...@moij.biz
Hi,
I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using
chan_ooh323 from asterisk-addons.
I am able to make a call
Have you got termination set correctly?
I have a B410P working with 2 x NT and 2 x TE ports successfully.
I had to turn the 100ohm termination on on the NT ports (even though I
have them set as PTP in mISDN.conf).
HTH
-- -Original Message-
-- From:
Philipp Kempgen schrieb:
Carlos Chavez schrieb:
Since 1.4.22 realtime status for sip peers seems to be broken. If I do
a sip show peers from the CLI I get this:
2001/2001 192.168.2.234D 5060 UNKNOWN
Cached RT
It is arbitrary which peers will
Carlos Chavez schrieb:
Since 1.4.22 realtime status for sip peers seems to be broken. If I do
a sip show peers from the CLI I get this:
2001/2001 192.168.2.234D 5060 UNKNOWN
Cached RT
It is arbitrary which peers will say OK and which will say
Tzafrir Cohen wrote:
On Wed, Jan 21, 2009 at 06:35:58PM -0600, troxlinux wrote:
Hi list, I install dahdi-linux successfully with the module of oslec
for the echo, but when I specify it in the system.conf the echo
canceller oslec it shows me errors:
DAHDI_ATTACH_ECHOCAN failed on channel 4:
Philipp Kempgen schrieb:
Philipp Kempgen schrieb:
Carlos Chavez schrieb:
Since 1.4.22 realtime status for sip peers seems to be broken. If I do
a sip show peers from the CLI I get this:
2001/2001 192.168.2.234D 5060 UNKNOWN
Cached RT
It is
Hi,
I'd like to know what's the most popular method for automatic fax/voice
detection for incoming calls on mISDN cards such as the B410P (hfcmulti).
I'm running:
kernel 2.6.17
misdn 1.1.3
asterisk 1.4.21.2
B410P card
I'm using iaxmodem and hylafax with asterisk (the setup works for zap
Greetings all,
I'm trying to connect to an ATT teleconference, but the
call is never marked as ANSWERED by asterisk and therefore won't bridge and
continue. The only work-around I've come up with so far is to dial like
this:
Exten = 744,1,Dial(Zap/g1,,p)
The private mode
Hello,
I am trying to connect an asterisk 1.6 to a trunking plate forme. With
asterisk 1.4.x I added to sip.conf a line asking for registration in the
form of:
register =
xx...@domain.com:Password:xx...@domain.comassword%3axx...@domain.com
@domain.com
Unfortunately, as you can see,
I'd try
xxx...@domain.com:Password:xxx...@domain.com
mailto:assword%3axx...@domain.com @domain.com
Or
'xx...@domain.com':Password: mailto:assword%3axx...@domain.com
'xx...@domain.com'@domain.com
_
From: asterisk-users-boun...@lists.digium.com
I have dahdi-linux-2.1.0.3 in centos 5.2 and the last version oslec svn
I have installed oslec and loaded, but it doesn't work me with dahdi
modinfo oslec
filename: /lib/modules/2.6.18-92.1.22.el5/kernel/net/ipv4/oslec.ko
description:Open Source Line Echo Canceller Zaptel Wrapper
In one of my center , its not taking root password.
Anyways to recover it ?
In other terms , I lost the control of server.
Any solution or re-installation is the only way left ?
I am using CentOS.
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What I have done in the past to set the password for root is to boot in
rescue mode and edit /etc/shadow setting the password to some know value
from another system.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: David @ULC ucoms2...@gmail.com
Reply-To: Asterisk
When GRUB comes up, select your boot entry and hit 'e' to edit. Then, arrow
down to your kernel line and hit 'e' again. At the very end of that line,
simply add the character '1'. Hit enter to save, then 'b' to boot. Your system
will now boot up in single user mode and drop you to a root shell
Or boot in single user
type passwd and done.
2009/1/22 Jim Dickenson dicken...@cfmc.com
What I have done in the past to set the password for root is to boot in
rescue mode and edit /etc/shadow setting the password to some know value
from another system.
--
Jim Dickenson
Jim Dickenson wrote:
What I have done in the past to set the password for root is to boot
in rescue mode and edit /etc/shadow setting the password to some know
value from another system.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
I personally prefer to chroot
Hi all,
Can we configure sip based outgoing fax on asterisk or we must need zap
channel attached with it?
Thanx in Advance.
Amir Shrestha
___
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asterisk-users mailing list
To
On Thu, Jan 22, 2009 at 11:32 AM, Max Brooks m...@legatio.com wrote:
Jim Dickenson wrote:
What I have done in the past to set the password for root is to boot
in rescue mode and edit /etc/shadow setting the password to some know
value from another system.
--
Jim Dickenson
Depends on codecs and frames you've selected. Zap use T.30 and SIP uses
T.38.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
amir...@namche.com
Sent: Friday, January 23, 2009 12:36 PM
To:
How many incoming calls will they support per line? You may find that
they support more than one incoming call per number.
Otherwise, get another provider.
Lyle
Alfred Monticello wrote:
I'm still stuck with this problem..Would appreciate any ideas anyone
might have on this one.
Thank you
On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala re...@supertec.com wrote:
Your service is still up and working,
Because Suzanne Bowen has better judgment than you.
You did charge back on the payment to us,
That is correct. There is $86 balance in my account I did not expect
to get back by
On Thu, Jan 22, 2009 at 08:28:18AM -0600, Danny Nicholas wrote:
Greetings all,
I'm trying to connect to an ATT teleconference, but the
call is never marked as ANSWERED by asterisk and therefore won't bridge and
continue. The only work-around I've come up with so far is
On Thu, Jan 22, 2009 at 11:54:59AM -0500, Matt Watson wrote:
For distros that do require a root password when booting single user mode,
your only real options have already been mentioned here...
1) boot from a CD, mount your partitions then:
Your other option is:
Add the boot option
This is *really* not the place for this...
On Thu, 22 Jan 2009, Andrew Joakimsen wrote:
On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala re...@supertec.com wrote:
Your service is still up and working,
Because Suzanne Bowen has better judgment than you.
You did charge back on the payment
Move it to the biz list and I am cool with it.
On Thu, Jan 22, 2009 at 12:51 PM, Jeff LaCoursiere j...@jeff.net wrote:
This is *really* not the place for this...
On Thu, 22 Jan 2009, Andrew Joakimsen wrote:
On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala re...@supertec.com wrote:
Your
Au contraire.
Jeff LaCoursiere wrote:
This is *really* not the place for this...
On Thu, 22 Jan 2009, Andrew Joakimsen wrote:
On Thu, Jan 22, 2009 at 19:34, Rehan Allah Wala re...@supertec.com wrote:
Your service is still up and working,
Because Suzanne Bowen has better judgment than
No, I really mean it. This list is busy enough with on-topic
conversations. If I wanted to hear rants about service providers I would
subscribe to *-biz. Good lord the net is *full* of crappy service
providers and frauds. The last thing we need is for members to believe it
is their duty
On Thu, 22 Jan 2009, troxlinux wrote:
I have dahdi-linux-2.1.0.3 in centos 5.2 and the last version oslec svn
I have installed oslec and loaded, but it doesn't work me with dahdi
modinfo oslec
filename: /lib/modules/2.6.18-92.1.22.el5/kernel/net/ipv4/oslec.ko
description:Open
True, better to keep it on the -biz list.
But there is such a thing as exceptional fraud.
I wouldn't have minded if someone posted about Bernie Madoff to
capital-users.
Jeff LaCoursiere wrote:
No, I really mean it. This list is busy enough with on-topic
conversations. If I wanted to hear
Randal Schwartz (FLOSS Weekly) joins us for this slightly OT
discussion about Open Source software and its role in all of our
lives. You're all welcome to join in as usual.
We will also be connected to a g722 bridge for more experimentation
with the merits of wide band audio.
This should give
Update on this - after reading a reply to another message from David Fire, I
put Answer in front of the dial. Now the dialplan is
Exten = 744,1,Answer()
Exten = 744,n,Dial(Zap/g1)
This may be the solution to those target numbers that don't generate the
proper tone or whatnot to allow * to know
There have been a number of answers provided. The one that was given to me
when I encountered this same problem was to boot a live CD, mount the root file
system and delete the password file which would force your normal distro boot
to request a new root password next time.
HOWEVER, the big
Vincent Li wrote:
On Thu, 22 Jan 2009, troxlinux wrote:
I have dahdi-linux-2.1.0.3 in centos 5.2 and the last version oslec svn
I have installed oslec and loaded, but it doesn't work me with dahdi
modinfo oslec
filename: /lib/modules/2.6.18-92.1.22.el5/kernel/net/ipv4/oslec.ko
When I boot I get this Error :
switchroot: mount failed: No such file or directory
kernel panic - not syncing: Attempted to kill init!
I tried :
1. Shut down the machine. (Ctrl+Alt+Del)
2. When it reboot and reach the CentOS boot up screen, then press any key to
go into a select menu. Then
Anyone using VicidialNow ?
I have documents for Vicidial scratch install but how to install step by
step Vicidialnow ?
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asterisk-users mailing list
To UNSUBSCRIBE or update options
The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 1.6.0.4, tagged as version 1.6.0.4-rc1. Release candidate
1.6.0.4-rc1 is available for immediate download at http://downloads.digium.com/.
This release candidate includes fixes for OS build compatibility,
Hi,
i have vicidial installed - and the company i have installed it for is
using it.
Take the VicidialNow install cd - install it - use it...
If you need to combine it with your own system - then follow the step by
step guid.
regards,
Wolfgang
David @ULC schrieb:
Anyone using VicidialNow
Can you configure the LAN port on the back of a 2102 to be bridged
rather than routed to the WAN port?
Cheers,
j
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asterisk-users mailing list
To UNSUBSCRIBE or update options
But I believe even after doing that , there are few setting and changes
required before we can start using it for production I guess...
On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.com wrote:
Anyone using VicidialNow ?
I have documents for Vicidial scratch install but how to
On Thu, Jan 22, 2009 at 3:11 PM, Jeff LaCoursiere j...@jeff.net wrote:
Can you configure the LAN port on the back of a 2102 to be bridged
rather than routed to the WAN port?
To my knowledge this is available on all Linksys ATA type devices that
offer both ports.
--
Kristian Kielhofner
Yes, it is available on the SPA2102 - you just login to the web interface,
goto the advanced section, then lan setup... its the very first option.
--
Matt
On Thu, Jan 22, 2009 at 3:29 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
On Thu, Jan 22, 2009 at 3:11 PM, Jeff
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808]:
On Thu, 22 Jan 2009, Wilton Helm wrote:
If some of your directories like /home and /user have separate mount
points, they don't have to get wiped out in the process.
If there is any reason to suspect a hack, re-installation is the only way.
I would replace the suspect drive and do a fresh
Hello all
I have used some low end cisco phones in the past and had no problem setting
up SIP on it.
But today, I have made a big mistake. Buying Cisco Conference phone without
even looking whether it supports SIP on not.
And yes it is the nice 7937G that I am talking about.
Damn this is
On Wed, 21 Jan 2009 19:02:01 -0500, Steve Totaro wrote:
Why not just get a softphone and use a USB soundcard or even the onboard
sound card as your ATA? Like a MagicJack and SJphone or Xlite or
whatever it is that works with it.
Please forgive my ignorance but I am not following you.
My
On Thu, 22 Jan 2009, Brian J. Murrell wrote:
On Wed, 21 Jan 2009 19:02:01 -0500, Steve Totaro wrote:
Why not just get a softphone and use a USB soundcard or even the onboard
sound card as your ATA? Like a MagicJack and SJphone or Xlite or
whatever it is that works with it.
Please forgive
The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really
is a great sounding phone. I have several customers with them as SCCP.
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/p
s8759/product_data_sheet0900aecd806e021a.html
From:
Hi everyone,
I have recently lost my Asterisk Admin job due to company's tight financial
situation. Now I am looking for another job and will appreciate any help.
I am good at Asterisk related VoIP stuff, LAN/WAN, IT, web, etc. and working
in this industry for more than 4 years now. Currently I
Yes I know too.
Is there anyway to make it work with asterisk without using Callmanager?
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons
(US)
Sent: Friday, January 23, 2009 5:10 AM
To:
Asterisk's Skinny support is very rudimentary and doesn't include the
CCM provisioning stuff.
Short answer - not really. Not unless you want to go through a *whole*
lot of work.
Sam Tam wrote:
Yes I know too.
Is there anyway to make it work with asterisk without using Callmanager?
Sam
On Thu, Jan 22, 2009 at 1:48 PM, Wilton Helm wh...@compuserve.com wrote:
There have been a number of answers provided. The one that was given to
me when I encountered this same problem was to boot a live CD, mount the
root file system and delete the password file which would force your
On Thu, Jan 22, 2009 at 1:52 PM, David @ULC ucoms2...@gmail.com wrote:
I tried :
1. Shut down the machine. (Ctrl+Alt+Del)
2. When it reboot and reach the CentOS boot up screen, then press any key
to go into a select menu. Then press e and navigate to the second line
grub.conf line
Hi, all. I want to execute a script, and return the value of said
(Python) script to the dialplan. I thought something like
exten = 1,1,Set(MyWorkingDir=System(/bin/pwd))
might work, but apparently not. I also looked into AGI stuff, but that
doesn't quite seem to be the right approach.
On Thu, 22 Jan 2009, Matt Watson wrote:
On Thu, Jan 22, 2009 at 1:52 PM, David @ULC ucoms2...@gmail.com wrote:
I tried :
1. Shut down the machine. (Ctrl+Alt+Del)
2. When it reboot and reach the CentOS boot up screen, then press any
key to go into a select menu. Then press e and navigate to
The guy who hacked me didn't seem too concerned about not being noticed. The
replacement ps would not allow me to kill any processes (including the ones he
was running). There was enough log information left that I could trace the
intrusion and even the ISPs hub it came from and I reported it
Does CENTOS have a rescue option?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, January 22, 2009 4:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Steve Edwards wrote:
On Thu, 22 Jan 2009, Wilton Helm wrote:
If some of your directories like /home and /user have separate mount
points, they don't have to get wiped out in the process.
If there is any reason to suspect a hack, re-installation is the only way.
I would replace
making sure to patch any holes through which the hacker might have come
In my case, I had been getting regular attacks through SSH for months, probably
100 a day (bots). Apparently after nine months of this, someone stumbled on to
my password which regrettably was composed of two dictionary
On Thursday 22 January 2009 16:16:33 Ken D'Ambrosio wrote:
Hi, all. I want to execute a script, and return the value of said
(Python) script to the dialplan. I thought something like
exten = 1,1,Set(MyWorkingDir=System(/bin/pwd))
might work, but apparently not. I also looked into AGI
hi
sorry i cant give you a job... but in the asterisk forum there is a jobs
section
http://forums.digium.com/
David
2009/1/22 Zeeshan Zakaria zisha...@gmail.com
Hi everyone,
I have recently lost my Asterisk Admin job due to company's tight financial
situation. Now I am looking for another
Folks,
First of all, this email is sent to -users and -biz, but please follow-
up to the -biz list only. I have set the reply-to, but I fear mailman
will strip it off ... Please don't flame me for posting to -users, I'm
just not sure who lives on -biz as the (signal/noise | net.kook |
On Thu, Jan 22, 2009 at 5:30 PM, Wilton Helm wh...@compuserve.com wrote:
Tripwire would be fine, if it had a baseline, but I don't think its any
good after the fact.
Correct - tripwire does need to be setup beforehand, and its not the most
simple thing to setup *properly*. After the fact...
On Thu, Jan 22, 2009 at 6:05 PM, Wilton Helm wh...@compuserve.com wrote:
making sure to patch any holes through which the hacker might have come
In my case, I had been getting regular attacks through SSH for months,
probably 100 a day (bots). Apparently after nine months of this, someone
If you are running a rpm based system you should be able to do ³rpm Va² and
see if anything has change from when all the stuff was installed.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Matt Watson m...@mattgwatson.ca
Reply-To: Asterisk Users Mailing List -
Brian J. Murrell wrote:
Slightly OT, but I'm wondering if anyone here has come across a soft
ATA. That is, software that will perform the functions of a basic POTS
line ATA on Linux with a zaptel driven card.
I have a Linux machine with a zaptel card in it and I want to have
another
On Thu, Jan 22, 2009 at 4:04 PM, Brian J. Murrell br...@interlinx.bc.ca wrote:
On Wed, 21 Jan 2009 19:02:01 -0500, Steve Totaro wrote:
Why not just get a softphone and use a USB soundcard or even the onboard
sound card as your ATA? Like a MagicJack and SJphone or Xlite or
whatever it is that
On Thu, Jan 22, 2009 at 7:58 PM, Steve Totaro
stot...@first-notification.com wrote:
On Thu, Jan 22, 2009 at 4:04 PM, Brian J. Murrell br...@interlinx.bc.ca
wrote:
On Wed, 21 Jan 2009 19:02:01 -0500, Steve Totaro wrote:
Why not just get a softphone and use a USB soundcard or even the onboard
Not sure if this is still valid - I used it on a project quite a while ago:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
PaulH
Ken D'Ambrosio wrote:
Hi, all. I want to execute a script, and return the value of said
(Python) script to the dialplan. I thought something like
Tilghman Lesher schrieb:
On Thursday 22 January 2009 16:16:33 Ken D'Ambrosio wrote:
Hi, all. I want to execute a script, and return the value of said
(Python) script to the dialplan. I thought something like
exten = 1,1,Set(MyWorkingDir=System(/bin/pwd))
might work, but apparently not. I
Hi David,
Thanks for your suggestion, I will give it a try at the Avaya at my
client's place.
Here is the set up for the failed call.
Caller - PSTN ---[Analog DID]--- Avaya ---[IP Trunk (H.323)]---
Asterisk ---[SIP]--- SIP Phone.
From my packet capture on the IP Trunk of the failed call, I
There's nothing special about analogue phones in China, they are fully
interchangable with analogue phones elsewhere... Perhaps you have a
configuration problem, or, hardware problem on the Rhino Channel Bank,
perhaps the ports are wired the wrong way and the phones care, perhaps
the
While I agree its better on the biz list, we do need some humor at the
end of business day.
On Thu, Jan 22, 2009 at 1:36 PM, Alex Balashov
abalas...@evaristesys.com wrote:
True, better to keep it on the -biz list.
But there is such a thing as exceptional fraud.
I wouldn't have minded if
Thanks for this link. Although this forum is not very active but I'll keep
checking it once in a while.
Zeeshan
On Thu, Jan 22, 2009 at 6:29 PM, David fire ddf...@gmail.com wrote:
hi
sorry i cant give you a job... but in the asterisk forum there is a jobs
section
http://forums.digium.com/
Google T38 - it's a big subject.
PaulH
amir...@namche.com wrote:
Hi all,
Can we configure sip based outgoing fax on asterisk or we must need zap
channel attached with it?
Thanx in Advance.
Amir Shrestha
___
-- Bandwidth and Colocation
2009/1/23 Lee, John (Sydney) john@compuware.com
There's nothing special about analogue phones in China, they are fully
interchangable with analogue phones elsewhere... Perhaps you have a
configuration problem, or, hardware problem on the Rhino Channel Bank,
perhaps the ports are
I've not used Rhino kit, but, that sounds like a firmware bug that
they
have a workaround for... With any luck it's very infrequent and
they'll
be releasing a fix once they've worked out the cause... Sorry I can't
help,
might be best to ask Rhino about the details of the problem...
The
On 1/22/09, David @ULC ucoms2...@gmail.com wrote:
But I believe even after doing that , there are few setting and changes
required before we can start using it for production I guess...
On Fri, Jan 23, 2009 at 12:27 AM, David @ULC ucoms2...@gmail.com wrote:
Anyone using VicidialNow ?
I
Looks like www.packet8.com http://www.packet8.com/ has been hacked
:-(
The phone service is offline as well.
Interesting to note that the whois is showing an update today but
doesn't look like details have changed.
Domain Name: PACKET8.NET
Registrar: REGISTER.COM, INC.
Whois
2009/1/23 Dean Collins d...@cognation.net
Looks like www.packet8.com has been hacked L
The phone service is offline as well.
Anyone else on this list using packet8?
Not using packet8, but, the website looks normal to me...
What are you seeing?
d
Nope it's going to a sedo advertising domain parking site.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
From:
On Fri, 23 Jan 2009, Dean Collins wrote:
Nope it's going to a sedo advertising domain parking site.
Looks like a real web site to me.
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867
Hey,
I am trying to work through a use case requirement where a user
listens to a some advertisement and then if at the end off it they
press a key they press a 1 key they get transfered to a pre-defined
number. I am using the asterisk java library at http://asterisk-java.org/
.
I was
On Fri, 23 Jan 2009, Dean Collins wrote:
Nope it's going to a sedo advertising domain parking site.
Looks like a normal website to me (from the UK). I'd check that the DNS
servers you're using haven't been hacked...
Gordon
___
-- Bandwidth and
On Fri, 23 Jan 2009 20:13:22 Peter Evans wrote:
On Fri, Jan 23, 2009 at 07:08:16AM +, Gordon Henderson wrote:
On Fri, 23 Jan 2009, Dean Collins wrote:
Nope it's going to a sedo advertising domain parking site.
Looks like a normal website to me (from the UK). I'd check that the DNS
And I assume no one know when they will have a SIP firmware for it too
right?
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Friday, January 23, 2009 5:44 AM
To: Asterisk Users Mailing
On 05:39, Fri 23 Jan 09, Sam Tam wrote:
Yes I know too.
Is there anyway to make it work with asterisk without using Callmanager?
Sam
Asterisk does have chan_skinny.
Featureset is not as good as CCM, but it's handling my phones and some
customers phones as well.
Check it out before returning
Well does it matter if the asterisk server is not located in the same
network?
I am willing to spend a bit of cash to get someone help me to set it up .
Since I need it quite done before end of this month
Sam
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From: asterisk-users-boun...@lists.digium.com
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