2012/3/13, resea...@businesstz.com :
> I am struggling to get the mac-addresses of IP phones that are connected
> to asterisk as the phone are in different VLAN with * and they were
> manually configured. I want to centralize their configuration using
> res_phoneprov or tftp
>
> I have tried nmap a
I understand you want to choose the easy way but I really think you should not
be lazy and go phone by phone and write down the Mac address. Of course if
that's ever possible...
For future ease of administering those phones, like if you want to do
provisioning, troubleshooting etc etc. Better g
On 15-03-12 01:54, Jan Blom wrote:
Hello,
Is anyone aware of an AMR-NB codec for Asterisk 10? The old patch
floating around for older versions of Asterisk doesn’t seem to work anymore.
The only patch I have seen is the one for 1.8 which is on sourceforge
(search for asterisk-amr). I did a qui
Hi,
Appreciate everyone for your valuable inputs. All these inputs provided by
you are really useful.
Thanks & Regards,
Amit Patkar
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Joi
I have a TDM410 with one FXO and one FXS. I've been running dahdi 2.5.0.2
without any problems. A couple of weeks ago I upgraded to 2.6.0 and found
that caller ID was no long working for me. All calls came in with a blank
caller id. I reverted back to 2.5.0.2 and everything was happy again. I
On Thu, Mar 15, 2012 at 10:04:56AM -0500, Chris Gentle wrote:
> I have a TDM410 with one FXO and one FXS. I've been running dahdi 2.5.0.2
> without any problems. A couple of weeks ago I upgraded to 2.6.0 and found
> that caller ID was no long working for me. All calls came in with a blank
> cal
Hi Guys,
I currently have an Asterisk 1.6.2.18 server running a patched (see
below) libpri 1.4.10.2 connected to a Toshiba Strata CTX670. All
external calls come in via the Strata and then are routed to the
Asterisk server over a single PRI link using Q931. This setup is
working and has been worki
On Thu, Mar 15, 2012 at 10:08 AM, Shaun Ruffell wrote:
> Hi Chris,
>
> I believe this is fixed in the head of the 2.6 branch. We're
> prepping a 2.6.0.1 release now...
>
Hey Shaun. Thanks for the quick reply. I applied the patch for the bug to
my 2.6.0 and it works fine. I've made five test
I'm wondering if any other Asterisk users have a recommendation for a reliable
SIP Trunk provider that supports Asterisk and offers decent support.
I've worked with Coredial, Broadvox, and Broadvoice and have had some bad
experiences with each of these providers.
Broadvoice offers low cost serv
I've had pretty good experience with VoicePulse.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jake Wicke
Sent: Thursday, March 15, 2012 10:46 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Reliable SIP Trunk Provid
You might check flowroute. We have been with them for over a year now and
have been spot on with service and support. www.flowroute.com and they
are one of the cheapest providers we have found for our needs.
Regards,
James Miller
Agent Black Web Hosting
"I see blindness, not as a disability, b
On 15/3/12 3:45 pm, Jake Wicke wrote:
I'm wondering if any other Asterisk users have a recommendation for a reliable
SIP Trunk provider that supports Asterisk and offers decent support.
You should probably let the list know what region/country you're in, as
you'll want to be as "close" (i.e.
With like 10 different ratesheets from 10 different providers, of which
many change their rates every few days, manually doing it in Excel is
too time consuming...
Am 15.03.2012 07:26, schrieb SamyGo:
could MS-Excel possibly be the easiest way to do that normalization !
just merge two rate sh
> I currently have an Asterisk 1.6.2.18 server running a patched (see
> below) libpri 1.4.10.2 connected to a Toshiba Strata CTX670. All
> external calls come in via the Strata and then are routed to the
> Asterisk server over a single PRI link using Q931. This setup is
> working and has been worki
I'm a fan of Vitelity. They are no-frills, but they work well for my very low
usage. I think their web portal is ugly, not all that intuitive, but it does
work. I've been with them since early 2006 for my few low usage DIDs.
-Original Message-
From: asterisk-users-boun...@lists.digiu
On Thu, Mar 15, 2012 at 9:02 AM, Chris Bagnall
wrote:
> On 15/3/12 3:45 pm, Jake Wicke wrote:
>
>> I'm wondering if any other Asterisk users have a recommendation for a
>> reliable SIP Trunk provider that supports Asterisk and offers decent
>> support.
>>
>
> You should probably let the list know
On Thursday 15 Mar 2012, Markus wrote:
> With like 10 different ratesheets from 10 different providers, of
> which many change their rates every few days, manually doing it in
> Excel is too time consuming...
Is it possible to get samples? I'd be interested in looking into
developing a script th
+1 for flowroute. very cheap and their support has been top notch when any
issues have come up
On Thu, Mar 15, 2012 at 12:15 PM, Carlos Alvarez wrote:
>
>
> On Thu, Mar 15, 2012 at 9:02 AM, Chris Bagnall > wrote:
>
>> On 15/3/12 3:45 pm, Jake Wicke wrote:
>>
>>> I'm wondering if any other Asteri
I have been using bandwidth.com since 2006 and have no problems at all.
They do not support t38 but have free local termination in a lot of US
cities. Tech support is good and they do support asterisk.
Jonn
On 03/15/2012 10:45 AM, Jake Wicke wrote:
I'm wondering if any other Asterisk users hav
Jake Wicke wrote:
> I'm wondering if any other Asterisk users have a recommendation for a
> reliable SIP Trunk provider that supports Asterisk and offers decent
> support.
>
> I've worked with Coredial, Broadvox, and Broadvoice and have had some bad
> experiences with each of these providers.
>
I'
Hello,
is there a way to disable a span for maintenance purpose (i.e. send yellow
alarm) ?
What would be the correct ioctl definition ? DAHDI_MAINT seems not to be the
right
candidate. Would DAHDI_SHUTDOWN send an alarm ?
Thanks
Hans
--
__
On Tuesday, March 13, 2012, Kevin P. Fleming wrote:
> On 03/13/2012 05:45 PM, Eric Wieling wrote:
>>
>> The faxdetect option is documented in the 1.8 sip.conf.sample.
>
> Right, I forgot about that. Now I've really confused things.
>
> /me heads back to his hole
>
It was actually added to chan si
On Thu, Mar 15, 2012 at 05:44:53PM +0100, Johann Steinwendtner wrote:
> is there a way to disable a span for maintenance purpose (i.e. send yellow
> alarm) ?
This could be a good feature to add to the dahdi_maint utility.
> What would be the correct ioctl definition ? DAHDI_MAINT seems not to be
On Thu,Mar 15 12:10:PM, Eric Wieling wrote:
> I'm a fan of Vitelity. They are no-frills, but they work well for my
> very low usage. I think their web portal is ugly, not all that
> intuitive, but it does work. I've been with them since early 2006
> for my few low usage DIDs.
>
+1 for Vitelit
www.didforslae.com have wide range of products to fit low usage to very
high usage. Dont want to put too much details here. Check it out let me
know if interested, since you are using I will help you waive activation
fee.
-Jai
On Thu, Mar 15, 2012 at 11:30 AM, Guy Gold wrote:
> On Thu,Mar 15 12
I use flowroute.com.
Intuitive GUI, cheap, and good customer service.
On Thu, Mar 15, 2012 at 1:30 PM, Guy Gold wrote:
> On Thu,Mar 15 12:10:PM, Eric Wieling wrote:
>
> > I'm a fan of Vitelity. They are no-frills, but they work well for my
> > very low usage. I think their web portal is ugl
Hello All,
I'm having issues with asterisk 1.8.4 dropping calls during transfer, and
transfer to park extension. We're using polycom soundpoint IP 650. when the
park button is hit the response is "i'm sorry not an extension" at the same
time number 7 appers on the lcd.
Thanks in advance.
-motty
> I'm having issues with asterisk 1.8.4 dropping calls during transfer,
> and
> transfer to park extension. We're using polycom soundpoint IP 650.
> when the
> park button is hit the response is "i'm sorry not an extension" at
> the same
> time number 7 appers on the lcd.
Please use a newer versio
On Wed, Mar 14, 2012 at 1:36 PM, Randall wrote:
> all works as expected only there is 1 extension that is trying to register
> with a wrong password causing fail2ban to block the IP address, normally
> that is ok behaviour but i have several extensions on that IP address.
>
>
First of all, white
Asterisk Project Security Advisory - AST-2012-003
Product Asterisk
Summary Stack Buffer Overflow in HTTP Manager
Nature of Advisory Exploitable Stack Buffer Overflow
Asterisk Project Security Advisory - AST-2012-002
ProductAsterisk
SummaryRemote Crash Vulnerability in Milliwatt Application
Nature of Advisory Exploitable Stack Buffer Overflow with locally
The Asterisk Development Team has announced security releases for Asterisk 1.4,
1.6.2, 1.8, and 10. The available security releases are released as versions
1.4.44, 1.6.2.23, 1.8.10.1, and 10.2.1.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/as
Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर):
On Thursday 15 Mar 2012, Markus wrote:
With like 10 different ratesheets from 10 different providers, of
which many change their rates every few days, manually doing it in
Excel is too time consuming...
Is it possible to get samples? I'd be
I would be more interested in a system where quality routes are tested with
different providers because rate really doesn't matter if a call can't be
placed or if a destination is a fake one. We have seen many fake
destinations with top tier providers but they had the best rates so the
strategy to
Our system just rolls over until it finds a carrier that will take it. Up to 30
different routes are supported, and rollover is pretty instantaneous in most
cases.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity and errors.
Alex Balashov - Principal
Evarist
>
> So, maybe a subscription service where a dialler system continuously tests
> routes with a list of 10 providers so that it's established which routes
> actually work and then allow that data to be downloaded for usage.
I think that it may not be humanly possible and also not possible to have
How about a central coop that manages the “normalized” rate sheet and
distributes it with “unknown” call quality metrics for each route. Coop members
report call quality for all calls/routes so the call quality metrics can be
updated in the rate sheet and distributed to members.
--Don
Don Kell
Good Idea but that means all the members of the coop use the same vendors
and it may not be suitable for servers having a bad network with a premium
quality provider and thus mark it as bad, whereas others are marking it as
Good. !!
On Fri, Mar 16, 2012 at 10:00 AM, Don Kelly wrote:
> How about
So the best fit is to create a piece of code which mixes providers and
destinations. 1- Sort those with at first rate and then as time passes
by(utilization) it rearranges the carriers according to quality metrics or
economical rates or success ratios. Like an AI system which enhances itself
with
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