similar to: Experiences with ATA model Octtel SP200SO

Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI> -- Executing Dial("SIP/16008-3d17", "SIP/16007&SIP/16006|20|tr") in new stack -- Called 16007 -- Called 16006

Hello list, I am currently doing a job for a summer camp. They would like to have several phones around the camp from which people can call in to the main office. It is an older campus and it is comprised of mostly old nungalow type housing. I need to install these phones several hundred yards from where the Asterisk server will be. The way thier telecom wires are currently set up is that they

I currently have 2 Cisco 7960's and 2 ATA 186's connected to asterisk. The 7960's work just fine for call waiting, but the ATA's dont. I cant seem to get the ATA's to use the call waiting feature, the calls just go straight to voicemail instead of prompting with the usual tone. Please help Chris -------------- next part -------------- An HTML attachment was scrubbed...

I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF

For those of you wanting to salvage your Cisco ATA-186 after inadvertent locking, or after recovering your devices from a vendor who has locked them, here is a rainy-day project for you: http://www.sst.com/downloads/datasheet/S71077.pdf The above document gives exact specifications on the 4mb flash EEPROM that stores all program and configuration data on the ATA-186 (aka Komodo.) If you

Anyone know of a company that makes a pass through ATA? By pass through I mean have an Ethernet switch built into the ATA, like most desktop phones have. All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN ports. I fooled around with DMZ etc...but it just doesn't work as well. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA's, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed...

I've got a user with Linksys ATA's for their analog phones. At random times during calls, the other party hears DTMF tones during the call. Is there a way to solve this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081120/3f1b4ff9/attachment.htm

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

Ok, I'm finally of a mindset to replace my now tired, aging Dell Inspiron 600m. I'm currently running Fedora Core on it, and am fairly happy with the software. Has anybody used CentOS on a Macbook Pro? What gotchas are there? (The first that comes to mind is the lack of a right mousebutton) Scouting on the Internet, I've seen that some people have trouble getting sound to work.

For all thos Asterisk users not on the FWD list, it works for me!: -----Original Message----- From: Free World Dialup - The Future of Dialing [mailto:FWD@LISTSERV.PULVER.COM] On Behalf Of Leonidas Piagkos Sent: donderdag 12 juni 2003 0:58 To: FWD@LISTSERV.PULVER.COM Subject: Re: [FWD] FWD losing Registration Hi Don, All you have to do with your ATA is to set the following parameters as :

Phil I have very good experience with the vegasteam ATA's devices.(you might also want to look @ sipura ATA's, since vegastream is doing an oem on there boxes) They support modem until v.90 speeds and faxes for g3. They are expensive, and again, work great and configure very easy joash ________________________________ From: asterisk-users-bounces@lists.digium.com

To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6 server i have tested a few T.38 capable ATA's: - Patton M-ATA - Grandstream HandyTone 486 - Fritz!Box 7170 I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also Asterisk 1.6.2.6 with Fax for Asterisk installed. These Asterisk servers are connected to a Cisco PGW 2200 + AS5400XM. Sending fax

Hello. I have a linux and two sip-ata's, a sipura 2002 and a GS ht-386. I also have three sipphone numbers. I can connect the atas to the sipphone accounts and I get a dial tone and I can call my house and it says, "Thank you for using SipPhone..." Using asterisk, I have the ata's registering to my computer and I register two sipphone numbers with my computer. When I

We are testing the innomedia ATA's to possibly replace our current line up of ATA's that we are using. Has anyone used their product? What is their track record on stability, voice quality, DTMF talkoff, T.38 Thanks Bryant ---------------------------------------- From: "Zeeshan Zakaria" <zishanov at gmail.com> Sent: Wednesday, October 13, 2010 10:41 AM To:

I read the thread about what IP phone is best for business deployment with great interest. Our need is slightly different however. We are deploying VoiP as a value-add with our high speed internet service and are having trouble finding the right SIP analog terminal adapter. In order to support people's existing phones and wiring we need to use an ATA. 1) The first priority is we want

Hi, Has anyone had any luck using modems on ata's other than with Cisco ATA-188's? I really don't have the money pay for the 188's as this is for my personal use. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041003/5ffde3f4/attachment.htm

I'm trying to figure out how Asterisk decides how often it will send SIP NOTIFY's to an ATA when a voicemail message is waiting for the user on the server. >From watching, it seems to be completely random. Sometimes 10 seconds apart, then 33 seconds, then 13 seconds, etc. Each time causes a "ring" on the ATA. I can change the ATA's behavior, but what I'd really

Does anyone know how the Linksys PAP2T ATA's can be mass provisioned? Documentation seems to be sketchy, even on the Linksys web site. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070607/3f90695c/attachment.htm

Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I don't make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? Kim C. Callis