Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

The mediatrix 4102s line kicks ass. On Jun 15, 2015 8:49 PM, "Matt Darnell" 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 latency. > > We are able

Re: [asterisk-users] Strange Polycom Issue

line of Cisco phones 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 wrote: > On Mon, Mar 9, 2015 at 9:40 AM, David Wessell wrote: > >> Welcome to our hell. >> >> We ran into this on VVX 3

Re: [asterisk-users] Strange Polycom Issue

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 wrote: > Welcome to our hell. > > We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We

Re: [asterisk-users] Strange Polycom Issue

or a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [image: Ringfree Communications, Inc] <http://rin

Re: [asterisk-users] Polycom DND + Intercom/Paging Override?

k.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- [image: Ringfree Communications, Inc] <http://ringfree.biz/> David Wessell / President 828-575-0030 x101/ da...@ringfree.biz

[asterisk-users] 401 Unathorized

- [image: Ringfree Communications, Inc] <http://ringfree.biz/> 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 con

[asterisk-users] Outlook CTI?

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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Polycom 4.1.6 BLF/URL Dialing

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 disa

Re: [asterisk-users] SPA112 Won't stay up

, 2014 at 12:54 PM, Mike Diehl 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 wrote: >> >> Is there another rou

Re: [asterisk-users] SPA112 Won't stay up

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 wrote: > I've got the registration period set to 15 minut

Re: [asterisk-users] CTI

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 wrote: > - Original Message - >> http://camrivox.com/products/fle

Re: [asterisk-users] CTI

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 wrote: > Hi people > > I'm just mailing to see what people are using for CTI solutions with > asterisk. Aslos, has anyone managed to integrate asterisk with Salesf

Re: [asterisk-users] Microsoft CRM Integration

http://www.camrivox.com/products/flexor-cti-dynamics-crm/ -- Ringfree Communications David Wessell 828-575-0030 x101 From: Steven Howes mailto:steve-li...@geekinter.net>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> D

[asterisk-users] AudioCodes MP-112

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 -- __

Re: [asterisk-users] Is Asternic.net out of business (Flash Operator, Call Center Stats)?

I have never known them to not reply quickly. Email me offlist and I will give you non generic email addresses. -- ringfree.biz supp...@ringfree.biz 828-575-0030 On Jun 17, 2013, at 8:14 PM, "Carlos Alvarez" wrote: > We have licensed both products and sent a support request on 6/11, with zero

[asterisk-users] AT&T uverse Motorolga nvg510

I have a client with AT&T 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

Re: [asterisk-users] Testing 911 call

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. -- Ringfree Communications David Wessell 828-575-0030 x101 From: James Miller mailto:paramedi

Re: [asterisk-users] multiple provider for incoming

o:asterisk-users@lists.digium.com>> Date: Tuesday, April 30, 2013 9:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Subject: Re: [asterisk-users] multiple provider for incoming On Tue, Apr 30, 2013 at 7:50 PM, David Wessell mailt

Re: [asterisk-users] multiple provider for incoming

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 mailto:mistral9...@hotmail.com>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.c

Re: [asterisk-users] Sip and the media path

etween themselves without involving Asterisk, but ones outside on the wan will be forced to talk directly to the Asterisk server for everything. You might also want to look at the nonat option of directmedia. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: David

[asterisk-users] Sip and the media path

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

Re: [asterisk-users] Dynamic Agents in a queue

e onto 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 mailto:kevin.lar...@pioneerballoon.com>> wrote: From: David Wessell mailto:da...@ringfree.biz>> To:Asterisk Users Mailing List - Non

[asterisk-users] Dynamic Agents in a queue

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 i

Re: [asterisk-users] [OT] Polycom IP450 Firmware Issues

Tim, What version are you on? There is a specific upgrade path for pre 3.3. Dw -- ringfree.biz Twitter: ringfreebiz 828-575-0030 On Dec 6, 2012, at 4:10 PM, "Tim Nelson" wrote: > I have a site with Polycom handsets on all the desks, mostly IP650s, some > IP550s, and some IP450s as well. > >

Re: [asterisk-users] Receptionist console (software)

Check out isymphony and fop2. -- ringfree.biz Twitter: ringfreebiz 828-575-0030 On Oct 3, 2012, at 5:49 AM, "James Mutuku" wrote: > Any recommendations I can use. I am looking on having software based > not a handset. > > -- > Best Regards, > James Mutuku Ndeti > Agile Systems Limited > +2547

Re: [asterisk-users] Running Asterisk on VMware ESX

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 On

Re: [asterisk-users] Asterisk and the media path

I really appreciate 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 wrote: > Hi Jared & Kevin, > > Thanks

Re: [asterisk-users] Asterisk and the media path

vin P. Fleming wrote: > On 05/21/2012 12:54 PM, David Wessell wrote: >> >> So I need directmedia set in sip.conf on the LCR trunk. >> >> 1) Do I need it in the individual trunk settings for each pbx? Or is >> in sip.conf enough? > > > You say 'in sip.co

Re: [asterisk-users] Asterisk and the media path

David On Mon, May 21, 2012 at 1:18 PM, Kevin P. Fleming 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. >

Re: [asterisk-users] Asterisk and the media path

id On Mon, May 21, 2012 at 11:22 AM, Kevin P. Fleming 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.. &

Re: [asterisk-users] Asterisk and the media path

ay 21, 2012 at 8:24 AM, SamyGo 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,

[asterisk-users] Asterisk and the media path

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 Trunk (Asterisk 1.6) > PBX (Asterisk 1.8). I am attempting to get the trunk to step out of the media stream. There i

[asterisk-users] Testing for media?

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 -- -- www.ringfree.biz 828-575-0030 -- __