No Kevin, we thank you for the fish, for sharing the knowledge and for
having the patience...
Good luck on your new journey!
P.S. As a token of our appreciation, just a word and we'll make the life of
the new Director of Software Technologies miserable!
2012/7/31 Kevin P. Fleming
> I've been w
Hello.
I've got a Cisco SPA303 that I'm trying to provision via http. I noticed that
this device looks very similar to a PAP2T, so I used that as a template for my
provisioning file.
However, the result is less than stellar. Line 1 registers and works.
However, lines 2 and 3 also register as
On Tuesday 31 Jul 2012, Kevin P. Fleming wrote:
> [snip]
> This is yet another incredibly exciting, career changing opportunity
> in my life, and I can't wait to see what it will bring. I'll be
> forever thankful for the opportunity that Digium and the Asterisk
> community provided me to learn, gro
On 1/08/2012, at 1:59 AM, "Kevin P. Fleming" wrote:
> I've been with Digium for just over seven years, and it's been an
> incredible experience that I wouldn't have traded for anything. When
> Mark Spencer invited me to visit Digium (and Huntsville) in early
> 2005, I could not have dreamed that
It's amazing what you can learn in a few days...
Having just found out that Queen Elizabeth has a great sense of humor, it has
now emerged that Kevin Fleming - a man who (both with and without his
moustache) has been an amazing contributor and influencer in the Asterisk
project is set to mov
On Tuesday 31 July 2012, bilal ghayyad wrote:
> Even at the web based configuration at the phone it self, I am not able to
> do reboot (there is no reboot button) and I can do this only from the
> Phone it self.
Real question: Why are you needing to reboot your phones?
You should never be gettin
Particularly, what virtualization software are you using?
On Tue, Jul 31, 2012 at 8:19 AM, Leandro Dardini wrote:
> Hello Bryant,
> it is nice to hear someone with different experience, so I am happy to
> know the "cloud" is indeed a feasible environment even for VoIP.
>
> Can you share with us
Is there an ETA on when this will show up on packages?
Thanks for the work!
EKG
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Hello Bryant,
it is nice to hear someone with different experience, so I am happy to know
the "cloud" is indeed a feasible environment even for VoIP.
Can you share with us some of your configuration magic? Like the cloud
service you are using, the power of each node and the load you are
experienci
However, there's no "reboot"
button in the web GUI of the phone.
I have no experience with the phone in question and so will make no
comment regarding the OP's original problem, but the absence of a
software reboot function in the web GUI seems to be a pretty major
oversight in my view.
I d
On 7/31/2012 7:22 AM, bilal ghayyad wrote:
Hi All;
Really it is miserable.
I bring 8 Digium Phone D40 and I used them with a customer, the voice quality
is bad internally (between the extension), there is no clearance at all ! We
are hearing the voice like another person.
The used codec is u
On Tue, Jul 31, 2012 at 10:59 AM, Kevin P. Fleming wrote:
> I've been with Digium for just over seven years, and it's been an
> incredible experience that I wouldn't have traded for anything. When
> Mark Spencer invited me to visit Digium (and Huntsville) in early
> 2005, I could not have dreamed
I've been with Digium for just over seven years, and it's been an
incredible experience that I wouldn't have traded for anything. When
Mark Spencer invited me to visit Digium (and Huntsville) in early
2005, I could not have dreamed that I'd end up working for such an
exciting, innovative company,
On 12-07-31 08:22 AM, bilal ghayyad wrote:
Hi All;
Really it is miserable.
I bring 8 Digium Phone D40 and I used them with a customer, the voice quality
is bad internally (between the extension), there is no clearance at all ! We
are hearing the voice like another person.
The used codec is u
On Mon, Jul 30, 2012 at 10:50:20AM +, Kamlesh Kumar wrote:
>
> when I issue 'make' command, below output comes. [root@localhost
> libpri-1.4.11.3]# make
> gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -O2 -MD
> -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_str
Kannan
I have to disagree with Leanrod. We are a hosted (cloud) PBX company we
successfully run our Multi-tenant systems in Virtual machines and have no
issues with them. It comes down to designing your virtual environment for
your target loads and then not exceeding them. This allows for fail
Hi All;
Really it is miserable.
I bring 8 Digium Phone D40 and I used them with a customer, the voice quality
is bad internally (between the extension), there is no clearance at all ! We
are hearing the voice like another person.
The used codec is ulaw.
The firmware version is: 1_1_0_0_48178
I have two setups with SIP hardware phones as extensions and POTS lines
as trunks. Internal SIP to SIP calls are crystal clear, but all calls
bridged to POTS have a significant amount of static noise. The problem
is that if I plug a POTS phone directly into the line, there is almost
no static n
hello,
i have strange problem with AGI (asterisk 1.8.10.0)
when i use Dial from dialplan everything is ok
when i dial from AGI script there is missing SIP Status 180 ringing and
183 session progress
any ideas?
DIAL without AGI
196.356479 10.0.0.193 -> 10.0.0.213 SIP/SDP Request: INVITE
sip:
On Mon, 2012-07-30 at 08:39 -0500, Matthew Jordan wrote:
>
> - Original Message -
> > From: "Ishfaq Malik"
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> >
> > Sent: Wednesday, July 18, 2012 9:58:47 AM
> > Subject: Re: [asterisk-users] Call recording and transfer is
On Mon, 2012-07-30 at 15:06 +0530, Kannan wrote:
> Hi
>
>
> I came across couple of pointers on the Internet regarding solutions
> available for providing hosted PBX service.
>
>
> 1. Multiple PBXs: Using separate hardware to host each PBX. Pretty
> straightforward, but no hosting company wants
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