search for: maroney

Does anyone know if the HandyTone 486 has the option to turn the two ethernet ports into either a switch/hub, or does it have to do NAT ? Thank you, Steve Maroney

...n echo canceller. I also noticed that its supports E&M Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does the phone systems have to be the same ? Thank you, Steve Maroney

I need to connect up to sixteen phones per building, I can use a cheap hub, but POE would be useful. Is there a cheap POE hub available? Everything I have seen has been expensive. Chris Mason

Hey guys, I was wondering what POE solutions are being used ? Ive done some searching on google and found that PowerDsine seems to be good brand. Any comments,suggestions, and experiences on POE hubs other POE products would be greatly appreciated. Thank you, Steve Maroney

...om solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a "loudspeaker" ? Thank you, Steve Maroney

...sk. I tried used the Call Me tool on fwdnet.net but I dont get any calls even though the Call Me tool says everything looks ok. Can someone call my FWD number and just leave me a message if i dont answer. FWD Number is 474538. My * box is configured to ring one of my extentions. Thank you, Steve Maroney

...ly replace a traditional PBXs with Asterisk/Linux but to provide a solution to needs such as a file serving, email serving, etc. Ive read the Success stories form voip-info.org but Im looking for a little more input on Askterisk Servers that host other network services as well. Thank you, Steve Maroney

...service. I would rather sign up with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is behind another Linux box serving as my nat/firewall. Does anybody think I will have a problem ? Should I stick to IAX and VoicePulse Connect or can I use VoicePulse with SIP ? Thank you, Steve Maroney

Hey guys, How much disk space is used by Asterisk to store voice mail for about 10 - 20 users/mailboxes Thank you, Steve Maroney

...Playing 'vm-youhave' (language 'en') -- Playing 'digits/9' (language 'en') So Im guessing its something to do with ADSI. So far, I only have this problem when checking voicemail, not for outgoing calls to another voip<-->pstn gateway. Thank you, Steve Maroney

...led SJPHONE ... no luck. I install x-lite and it does it too. I removed XP SP2, and also updated my sound drivers .. no luck. Any one ever had this heppen to them ? Please Help. I also tried other sip clinets on another pc with no problem so it doesn't seem to be the server. Thank you, Steve Maroney

...l tone after the last caller gets added to the database. When I look in my sounds directory i see a few files such as to-blklist-last-caller.gsm and enter-num-blacklist.gsm. Is there a more developed interface to (un)blacklisting numbers or do I have create one in my dialplan ? Thank you, Steve Maroney

I easily get confused when try to undertstand FXO & FXS ports. Is it possible to use an ATA to connect to a TDM400 card. If so, would I use FXO modules or FXS modules ? My goal is to connect my asterisk server to Vonage (via the ATA they send me) so I can use thier standard plan and do with out the Softphone account feature that only allows a few hundred minutes talk time. Thank you, Steve

Why does this person have my e-mail address ? ----- Original Message ----- From: <markogift@astriholics.org> To: <hackerwacker@cybermesa.com> Sent: Tuesday, November 23, 2004 1:13 PM Subject: Gift for Mark Spencer > Hello everyone! > > We have been thinking about something that we could do for Mark > Spencer as a holiday gift. We have decided to try to orgranize a

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

Does anyone know of any plans to add an intercom/all-page feature in *? The few SIP phones that have auto-answer capability would be better if Asterisk could broadcast one leg of a channel to many legs at one time. Thank you, Steve Maroney

Hello list, I've posted my problem on BSD list and i still have the problem. The remote side receives the call , but there's no voice on the call. I tried everything about possible NAT problems .. but ther're on same net. My platform: FreeBSD 5.2.1-Release Asterisk 1.0-RC2 soft phones : X-Lite >>>> -- Executing Dial("SIP/1260-a7ae", "SIP/1262|20")

I have a couple of questions on the zapbarge: 1) zapbarge asks for a channel - how would a manager know what channel to enter ? Is there any way of being able to enter an extension number instead ? I know that you can get the information from the manager interface, but I wouldn't want to give my users access to this, or have to install / write a system just to get an extension number from a

Hi guys, For the first time, I'm attempting to run asterisk as a non-root user for all of the obvious reasons. I'm attempting this with asterisk-1.0-RC2, based on the fairly straightforward directions found here: http://voip-info.org/tiki-index.php?page=Asterisk+non-root The only problem I can't get figured out is my mpg123 processes not being spawned properly. There's

First, I am surprised at IAX... My asterisk server is behind a firewall, I am behind a firewall, and my iax client connected... Voodoo??? :) But, the firewall has one unblocked port, which I think I should set to the IAX, so the magic won't be necessary, and it makes sense to think that there will be less latency without the "magic layer". But, when I set port=myport in the