The mediatrix 4102s line kicks ass.
On Jun 15, 2015 8:49 PM, Matt Darnell mattdarn...@gmail.com wrote:
In the past we have used Adtran Atlas 550's to break out FXS ports for
devices like modems. The great thing about the 550 is that internally it
is all TDM so there is absolutely zero
coming out in May.. If they're any
good we'll strongly consider those...
dw
On Mon, Mar 9, 2015 at 10:55 PM, Ryan Wagoner rswago...@gmail.com wrote:
On Mon, Mar 9, 2015 at 9:40 AM, David Wessell da...@ringfree.biz wrote:
Welcome to our hell.
We ran into this on VVX 300 and 400 phones running UCS
://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Ringfree
I'll add that it appears to happen when you have users in a ring group or
call queue and BLF is being used in some capacity..
dw
On Mon, Mar 9, 2015 at 9:40 AM, David Wessell da...@ringfree.biz wrote:
Welcome to our hell.
We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We
/listinfo/asterisk-users
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David Wessell / President
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Ringfree Communications, Inc Office: 828-575-0030 / Fax: 888-243-7830
PO BOX 1994 Hendersonville, NC 28793
http://ringfree.biz
This e-mail
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David Wessell / President
828-575-0030 x101/ da...@ringfree.biz
Ringfree Communications, Inc Office: 828-575-0030 / Fax: 888-243-7830
PO BOX 1994 Hendersonville, NC 28793
http://ringfree.biz
This e-mail message may contain confidential or legally privileged
information and is intended only for the use
Are there any quality Outlook integrations for asterisk out there? The
closest I'm finding is at http://camrivox.com and they don't support
Outlook 2013.
dw
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I've just deployed several VVX 600's with the Color Expansion Module.
And I'm having a minor issue with them.
Intermittently when a call comes into a ring group the user is
presented with the call pickup option associated with a BLF entry. Not
the normal answer/reject option.
I've explicitly
Is there another router in the mix? Put the cable modem in bridge mode and
attAch a real router.
http://401stblow.wordpress.com/2012/10/21/fixing-time-warner-cable-ubee-modem-connectivity-issues/
On Thursday, February 6, 2014, Mike Diehl mdiehlena...@gmail.com wrote:
I've got the registration
PM, Mike Diehl mdiehlena...@gmail.com wrote:
Unfortunately, we plug straight into the Ubee and the ISP will not support
any other modem.
GRRr..
Mike.
On Thu, Feb 6, 2014 at 12:34 PM, David Wessell da...@ringfree.biz wrote:
Is there another router in the mix? Put the cable
http://camrivox.com/products/flexor-cti-salesforce/
We've used this for a few clients.
On Fri, Jan 10, 2014 at 6:33 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi people
I'm just mailing to see what people are using for CTI solutions with
asterisk. Aslos, has anyone managed to integrate
No major issues. They're always very responsive. I'd get a demo from
them for the client and make sure that the feature set is a match. But
I always say that with 3rd party apps.
On Fri, Jan 10, 2014 at 10:39 AM, Tim Nelson tnel...@rockbochs.com wrote:
- Original Message -
http://www.camrivox.com/products/flexor-cti-dynamics-crm/
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David Wessell
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From: Steven Howes steve-li...@geekinter.netmailto:steve-li...@geekinter.net
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Does anyone have experience setting up an AudoCodes MP-X with an asterisk
(FreePBX based) system? I would be willing to pay a reasonable amount for
assistance with the MP-X device. I have remote access setup, so no one should
have to leave their comfy chair..
Thanks
David
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I have never known them to not reply quickly. Email me offlist and I will give
you non generic email addresses.
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supp...@ringfree.biz
828-575-0030
On Jun 17, 2013, at 8:14 PM, Carlos Alvarez car...@televolve.com wrote:
We have licensed both products and sent a support request
I have a client with ATT uverse and the modem mentioned above. They are
connecting to an offsite asterisk server running 1.8.
All functionality seems to be fine except for occasionally they are unable to
pick up calls. We do not have this problem from any other location.
Typically with a dsl
Quite a few SIP providers will have 911 testing functionality. Our main 911
provider lets you dial 933. Than they read back to you the address information
that is transmitted with the 911 call.
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Ringfree Communications
David Wessell
828-575-0030 x101
From: James Miller paramedi
Hi Matt,
You can't have multiple providers for inbound traffic. You can have multiple
providers for outbound traffic though.
Thanks
David
From: Matt Hamilton mistral9...@hotmail.commailto:mistral9...@hotmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
On Tue, Apr 30, 2013 at 7:50 PM, David Wessell
da...@ringfree.bizmailto:da...@ringfree.biz wrote:
Hi Matt,
You can't have multiple providers for inbound traffic. You can have multiple
providers for outbound traffic though.
Thanks
David
David,
I'm not sure where you got this information, but it's
We're running asterisk 1.8 in the DC on a public IP address.
Connecting to it are about 200 phones behind a LAN in a remote location.
Is there a way to reliably keep asterisk out of the media stream on internal
calls inside that LAN? All phones are Polycom Soundpoint phones.
Asterisk would say
Analyst - Pioneer Balloon - Ph: 316-688-8208
From:David Wessell da...@ringfree.bizmailto:da...@ringfree.biz
To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com,
Date:04/25/2013 07:33 AM
Subject
the next penalty level.
Is this the same behavior that you have seen?
Thanks
David
On Feb 28, 2013, at 5:55 PM, Kevin Larsen
kevin.lar...@pioneerballoon.commailto:kevin.lar...@pioneerballoon.com wrote:
From:David Wessell da...@ringfree.bizmailto:da...@ringfree.biz
To:Asterisk Users
Hi,
We have a queue running with dynamic agents in asterisk 1.8.12.0 and FreePBX
2.10.
We are using the linear ring style.
Calls are going to the agents in the order in which they log in.
Is there a way to send calls to an agents in a specific listed order and not in
the order that they log
Tim,
What version are you on? There is a specific upgrade path for pre 3.3.
Dw
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ringfree.biz
Twitter: ringfreebiz
828-575-0030
On Dec 6, 2012, at 4:10 PM, Tim Nelson tnel...@rockbochs.com wrote:
I have a site with Polycom handsets on all the desks, mostly IP650s, some
IP550s, and some
Check out isymphony and fop2.
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Twitter: ringfreebiz
828-575-0030
On Oct 3, 2012, at 5:49 AM, James Mutuku listmut...@gmail.com wrote:
Any recommendations I can use. I am looking on having software based
not a handset.
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Best Regards,
James Mutuku Ndeti
Agile Systems
We virtualize every asterisk install, and have achieved density levels of
80MB RAM per install of asterisk. We do it all day, every day.
As Chris wrote if you're putting it on shared hardware that you don't
control, just don't. If you control all of the hardware it's very doable.
Thanks
David
the suggestions.
Screenshot:
http://dl.dropbox.com/u/4156401/Screenshot%20from%202012-05-23%2007%3A39%3A51.png
pcap: http://dl.dropbox.com/u/4156401/trace3000.pcap
Thanks
David
On Wed, May 23, 2012 at 7:41 AM, David Wessell da...@ringfree.biz wrote:
Hi Jared Kevin,
Thanks for taking the time
:24 AM, SamyGo govoi...@gmail.com wrote:
Hi,
Can you check if there is any transcoding involved with these calls, or
maybe some NAT handling needs to be done by asterisk so it's not stepping
out of the media-path !?
Regards,
Sammy
On Mon, May 21, 2012 at 5:03 PM, David Wessell da
, May 21, 2012 at 11:22 AM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/21/2012 07:03 AM, David Wessell wrote:
I am attempting to get an asterisk server to step out of the media
path, but am running into a brick wall. Can someone assist? Here's my
setup..
Ultimate SIP Provider --- LCR
David
On Mon, May 21, 2012 at 1:18 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/21/2012 11:46 AM, David Wessell wrote:
Hi Kevin,
Thank you. Here's the requested information.
1) The Trunk is running 1.6.2.9. Also it's running a2billing.
2) The PBX is running asterisk 1.8.12.0 along
I'm in the process of setting up an asterisk box that will stand
between PBX's and the SIP providers. So a trunking server.
How can I 'test' to see if this trunking server is stepping out of the
media path during calls?
Thanks
David
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