It appears as though SELinux is preventing you from moving forward.
Perform the following to disable SELinux.
cd /etc/selinux
vi config
change "enabled" to "disabled"
write your changes
reboot
Bryan M. Johns
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
Suppo
We have multiple installs that tested-out at nearly concurrent 400 SIP
channels on a Dell 2950 with 2Xquad core at 1.6 Ghz, 16 GB of RAM.
Bryan M. Johns
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
Support: [EMAIL PROTECTED]
http://www.sheltonjohns.com
On Feb 8, 2008, at 5:09 AM
You aren't including the file extension when referencing the graphic name, are
you? If so, that would be the problem. You might also want to try loading the
parameters to the fields for the 650 also.
Just a couple of ideas.
Bryan M. Johns
Partner
Shelton | Johns
Office: 678.248
As a pure SIP solution, we have switched as many as 120 call paths through a VM
on a lightly populated host.
Bryan M. Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: "WipeOut" <[EMAIL PR
Make certain that selinux, iptables and ip6tables are disabled and off.
Bryan M. Johns
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
Support: [EMAIL PROTECTED]
http://www.sheltonjohns.com
On Dec 2, 2007, at 3:18 PM, James Cox wrote:
> I have asterisk up and running on a fed
servers, two voicemail servers, one
conferencing server, etc.
Just some ideas. Best of luck to you!
Bryan M. Johns
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
Support: [EMAIL PROTECTED]
http://www.sheltonjohns.com
On Dec 27, 2007, at 11:33 AM, Jesse Molina wrote:
>
> Anyon
Bob,
We are on a similar assignment right now. Please contact me off-list if you
would like to discuss how we might be helpful.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From
You can use an inexpensive PC with a sound card. Install Asterisk on
it and set an extension that calls /dev/dsp. This will send audio
out the speaker port on the sound card.
Setup a trunk between this unit and your primary Asterisk server and
you should be in business.
Bryan M. Johns
Stephen,
Thanks for the heads-up on the cab ride from Phoenix to the event. I did not
know it was that far. I will be coming in Wednesday morning and I may take the
same route you are considering.
Anybody coming in Wednesday morning that wants to split fare?
Bryan M. Johns
Partner
Insert your offset into this line:
tcpIpApp.sntp.gmtOffset=""
eg - EST (GMT -5) = -18000
Bryan M. Johns
Shelton | Johns
678.248.2637 Office
678.810.0730 Direct
678.303.3424 Fax
Support: [EMAIL PROTECTED]
http://www.sheltonjohns.com
On Nov 18, 2008, at 5:46 PM, Doug Smith wrote:
Tried
I recommend looking through the settings available in these files and
maybe doing some research on the config of particular PolyCom models
at http://www.voip-info.org. Backup your original configs before
making edits, though ;-).
I hope this is helpful.
Bryan M. Johns
Partner
Shelton | J
I recommend the hitachi wifi phones for use with asterisk.
Bryan M. Johns
Partner
Shelton Johns Technology Group
Office: (678) 248-2637 X: 1500
Direct: (678) 229-1809
http://www.sheltonjohns.com
**Sent from my mobile phone**
-Original Message-
From: "Steven" <[EMAIL P
The device config for the Snom 360 needs to be set to adhoc mode. If you are
not comfortable with hand-configuration of the extensions file, take a look at
freepbx as a tool to assist you.
Thanks,
Bryan M. Johns
Partner
Shelton Johns Technology Group
Office: (678) 248-2637 X: 1500
Direct: (678
I guess I misunderstood your issue, Fred.
Have a great New Years.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Dec 30, 2006, at 8:59 AM, Asterisk
Most SIP phones handle this functionality by recognizing numbers from
speed dial or address book entries in the phone itself. I believe
that the PolyCom SIP phones do this (IP430, IP501, IP601, IP650).
I hope that this is helpful.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
Ray,
Have you considered using a VM architecture?
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Jan 5, 2007, at 5:17 PM, Ray Jackson wrote:
Hi all,
I
-alone server and by downgrading
all DomUs to Asterisk 1.2.14.
Our entire architecture is Fedora 6, by the way. DomU is 32bit and
all DomUs are run on a single, large 64bit server platform.
I hope this is helpful.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x
I wish had some pearl of wisdom here, but I don't. I will simply
share my sympathy.
Sounds like an ESU situation to me.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
Exactly.
ESU = Equipment Superior to Users
;-)
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Jan 10, 2007, at 12:47 PM, Alexander Lopez wrote:
More
Are you using tftp or ftp provisioning? If so, check your server
declaration in sip.cfg in your polycom configs directory.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http
I ran into this problem with an early batch of IP650s. Polycom's
firmware version 2.0.3b made this issue go away.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
Jason,
Email me off-list and I will ship you a pack of usable configs.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Jan 26, 2007, at 3:48 PM
voice quality.
I hope this is helpful.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Feb 6, 2007, at 8:09 PM, Jim Duda wrote:
I'm struggling to g
Assuming you are using a central provisioning server, check your
{MAC}-directory.xml file. It contains the ordering that you are
looking for.
I hope this helps.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
tion capability is more broad than
any other phone we have worked with.
I hope this information is helpful.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
iaxtel: 700:248:2637 x:1500
http://www.s
What asterisk version?
Bryan M. Johns
Partner
Shelton Johns Technology Group
Office: 678.248.2637
Direct: 678.229.1809
http://www.sheltonjohns.com
**Sent from my mobile phone**
-Original Message-
From: "Jason Wolfe" <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
DST rules can be found by searching the sip.cfgg file for "SNTP".
You will find a cluster of time parameters, including the month and
day upon which to change DST.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
i
you want more details, please feel free to contact me off-list.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Mar 22, 2007, at 3:50 PM, Chris Bagnall wrote:
Greetings
wn on network provisioning the Polycom:
http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501
While it may not be practical to go this route for a single desk
handset, it can be a life-saver in a larger network rollout.
I hope that this helpful.
Bryan M. Johns
Partner
Shelton |
Is your carrier delivering service via a TDM circuit?
It has been our experience that you will get far more reliable fax
performance via the method you describe (analog device terminated to
a port on a FXS line card) than attempting to use an ATA on the LAN.
However, if your carrier is a S
Might want to look into you libtiff version and / or the presence of tiff2pdf.
Just a guess.
Bryan Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: "Sahil Gupta" <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Sent:
No relevant experience with OpenVZ, but plenty with Xen if you would find that
interesting.
Bryan Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: "Voip Asterisk" <[EMAIL PROTECTED]>
To: "Asterisk Use
What you are describing is only available in a guest domain if your CPU(s)
support hardware virtualization. If they do, however, this configuration is
pretty straight forward.
Xen as a virtualizing solution ships in a well-documented format in the Fedora
6 distribution. If you would prefer to
Install zaptel and only enable the ztdummy module. As long as you are not
running in a VM, this will supply you the timing that you are looking for.
Bryan Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
Fro
Have a look at these guys: http://www.vitelty.com
I have had good success with their service (particularly with porting).
Bryan Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: "Salvatore Giudice" <[EMAIL PRO
Might want to confirm what server address you have declared in your sip.cfg
file (assuming you are using network provisioning for the phones).
Bryan Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: "Eric \"Man
What version are you running? Anything creative like VMs or other unique
configurations in use?
Bryan Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: "Wayne Jensen" <[EMAIL PROTECTED]>
To: "Asterisk Users Ma
We saw this behavior early in the 1.4 releases and shelved 1.4 upgrades for the
time being. The behavior that we saw was similar to what you describe.
Bryan Johns
Partner
Shelton | Johns
Office: 678.248.2637
FindMe: 678.229.1809
http://www.sheltonjohns.com
- Original Message -
From: "T
With Polycom phones, you should steer clear of headsets with in-line
amplifiers. We have found these to introduce electrical hum into the
audio streams.
Just an FYI.
Thanks,
Bryan Johns
Partner
Shelton | Johns
1805 Old Alabama Road
Suite 200
Roswell, GA 30076
USA
Office: 678.248.2637
Find
I've experiencing some difficulty passing inbound calls from the PSTN, through a large Asterisk switch and down our network to a Cisco 1751 router. This router has 4 FXS ports and is running IOS 12.3.
Outbound dialing from phones on the FXS ports of the router works flawlessly, but inbound ca
On Thu, 2005-06-16 at 23:24 -0400, Nate Kapi wrote:
When using the dial command and the D option to send DTMF digits when
the channel is answered, is there a way to allow for some dead air,
and then send more DTMF digits? I would like to automate a call, and
it requires entry of a few short d
Bjorn,
Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal
It has a flash-based panel that will give you what you are looking for.
Bryan M. Johns
One Ring Networks
300 West Wieuca Road, NE
Building One
Suite 205
Atlanta, GA 30342
404.303.9900 x: 104
Bjorn,
Take a look at the Asterisk Management Portal at http://sourceforge.net/projects/amportal
It has a flash-based panel that will give you what you are looking for.
Bryan M. Johns
One Ring Networks
300 West Wieuca Road, NE
Building One
Suite 205
Atlanta, GA 30342
404.303.9900 x: 104
What DTMF mode are you using?
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