similar to: Sipura SPA-3000 Beta

Hi All, Last week I started hearing a huge amount of local end echo on incomming calls. I am using a Sipura SPA-3000 as my FXO connected to an SBC POTS line. Echo cancellation is enabled in the SPA firmware. As a test I switched to a Digium X100 card the still lives in my server but the echo was about the same. I have both Polycom IP600 and SNOM 200 phone, which both hear the echo. I'm

All of a sudden my list traffic appears to have dropped to a few messages/day the past few days. I anyone else seeing this as well? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com "...I believe in love, its all we've

Anyone here have experience with these devices? They would ppear to be an affordable alternative to multiple X100Ps. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com "Kick at the darkness 'till it bleeds daylight" - Bruce

OK. So it would appear that my quest for FXO adapters unconvers more, and certainly more mature, H.323 based devices...not so many SIP devices. What would be the benefits of SIP over H.323 for a small office * server? All I need to do is bring 4 POTS lines into * with Caller ID, make outgoing local calls reliably without undo echo. FWIW, my * server is Fedora Core 1 on AMD XP2500 with 512 MB

Hi All, Is this working for any of you? We're supposed to be able to dial FWD accounts through IAXTel by dialing 170099+FWD number. I don't ever seem to be able to do this even though I can dial 800 numbers through IAXTel. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once

Hello All, I have a small * server in my home office with several IP phones. The system is not fully in service yet as I'm still hunting for a cost effective FXO adapter that I can rely upon for my two primary PSTNs. That said, I'd like to move it into service for another application...which brings up a question. I'd really like to stop making international calls from my cell phone

Hello All, I've decided to go ahead and plug my fax machine into one half of a Sipura SPA-2000. I've Google the list archive for examples of default fax contexts but not come up witha clearly working result. Can someone here please send me a config fragment that theyt know to work? Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr. Product

Can anyone here help me in getting connected to Clearpath? They have supposedly setup a DID and 800 number for me but not provided login info. They're really hard to reach. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com "If

Have the Sayson IP phon started to deliver yet? I'm thinking about two new phones for my office and considering the Sayson 480i and Zultys 4x4. Would also consider the Virbiage phone if it becomes available. I have Snom 200s and a Pingtel phone at the moment. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

Hello All, I'm not certain that I have my firewall setup to perform traffic shaping correctly for *. I'm using a m0n0wall running on a Soekris 4501 platform. I have traffic shaping set to provide IAX port 4569 with priority outbound access from LAN to WAN. All other outbound access is lower priority. This should in theory accomodate my ADSL connection which is 1.5 Mbps inbound but only

...my VoicePulse Connect account is timing out on its login requests. Was working fine a hour ago. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262 "Can't you see it all makes perfect

That about says it. I upgraded from 1/24/2004 to the most recent CVS and my calls through VoicePulse Connect over IAX2 are simply unusable. What's the story on this? Is there some setup that I'm missing? I had jitterbuffer=no, changed it to yess with no apparent effect. Michael FWIW, have attemped to open an account with Nufone 3 days ago. It has yet to be activated. -- Michael Graves

Hi All, A friend has asked me to get a new Zultys 4x5 working with his * server. I've been over the web gui but don't see where all the registration info gets entered in the phone. Is there someone on-list who has Zultys phones and can advise me as to their setup? The manufacturers docs are useless. Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr.

If that was possible, that would make my life easier as well :) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Graves Sent: Tuesday, August 31, 2004 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom SoundPoint... Gains - Whichis for speakerphone

Hi All, Does anyone on-list have experience with the Zultys 4x4 and 4x5 phones with AES encryption? Does this work well with *? I imagine that it would work as long as * doesn't have to decode the stream. What are the implications for calls going to voicemail or MeetMe? I'd pondering installing the 4x5, especially if Zultys can make the on-board FXO port available to * as well as a

John, By chance do you know how to set a default ringer? What I have done is the following: <DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="7" se.rt.1.callWait="6" se.rt.1.mod="1"/> As you can see, I want 7 to be the default ringer for line 1... For some reason, it doesn't take these changes.

Please forgive me if this is a silly question. I've been following this thread in the hope that I could put my * server and snom 200 into full-time service very soon. I need to find out how to have the lines configured so that it does not return a busy reply when only one call instances is engaged. Am I supposed to create multiple extensions on my asterisk dialplan to reflect the 5 call

Anyone have one of these yet? http://www.voxilla.com/shop/index.php?action=item&id=38 -Dan

Hello All, I'm about 1/2 way through building an * system for home office. I have the server built, * installed, but I'm waiting for a Grandstream phone to arrive to have a reliable client. In the mean time I have a few short questions. 1. I have the X-Pro soft phone client. It's presently configured for FWD. What settings would I need in it to get it to use the * server? I have *