similar to: Lower cost router suitable for VoIP?

Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we

They say on their website that they allow you to use your own device provided you give them the MAC address. Has anyone tried using * with it? Looks like they have quite a few rate centers and also phone support... Their website is horrible though... Just wondering, it'd be good to get user experiences from different providers other than IconnectHere and BroadVoice... -Chris

Back to the ACTUAL TOPIC of this thread... This phone looks kinda nice, where can one get hold of it? How about it's * compatibility? I realize that it says it does things like 3-way conference and attended transfers, but how about in *? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

don't quote me on this but I believe the earlier assumtion is correction. I think you need to have RTP going bothways otherwise the call will disconnect. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Shaw Sent: Friday, August 13, 2004 12:40 PM To: asterisk-users@lists.digium.com Subject: Re:

I have really tried to do my best googling and wiki-reading before asking this question. I couldn't find the answers there so I throw myself at the mercy of the list... I get excellent quality for SIP -> PSTN and PSTN -> SIP calls, however when I or anyone else calls from PSTN -> * the voice menus are oftentimes very choppy. Sometimes they are absolutely perfect and I cannot tell

I know this must have been asked before, but I was just wondering, the manual says it can do attended transfers, has anyone gotten this to work successfully? How did they do it? Is it possible to do attended transfers with the 'T' dial option? If so, how? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

Now that most of you have worked overtime to show why most people are continually pissed at Nix Users (all except two of course). The problem I can see is the downright technosnobbery involved. There is nothing wrong with Linux. I play around with RH9 and FreeBSD and find that most things run fine. But you get into a problem where it keeps asking for the same blamed libraries over and over on

Hello, I'm searching for a router for our company. Does anybody has a suggestion for a router with a SIP Application Layer Gateway and good working QoS (Upstream AND Downstream). Regards Bastian

Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are

Hi Team, Just, I would like to know, is there any unlocked device(wireless router with built in FXS port) for home users which are connected Asterisk based VOIP service. I have looked products from Linksys and D-link etc. But all these products are bundled with VOIP Service providers (vonage, lindo and at&t) . Please sugest if any products avaiable in market. Thanks, Karun

Has anyone configured the Zoom V3 to connect with Asterisk? I bought one at Fry in San Diego for $99 bucks. I can't get it to register with Asterisk.. Thanks, David http://www.zoom.com/products/voip_products.html

On Tue, 2005-05-31 at 13:38 -0400, Jared Mauch wrote: > On Tue, May 31, 2005 at 12:30:07PM -0500, John Lange wrote: > > I am not a Cisco person; so the question is, is it possible to have one > > of the following: > > > > 1) Have the Cisco authenticate (register) as a SIP client to the > > Asterisk server. This allows me to place the Cisco in its own context. >

Hi All: I'd like to test a pure VoIP call center set up under Asterisk, Can I use existing IP routers to get VoIP traffic from service provider to Asterisk with good quality of voice? In other words, do I have to do any hardware upgrade to make VoIP work in existing enterprise environement, we have 10g Ethernet LAN? Many thanks, Tielin

Hi everybody, What is a proper setup for a medium size business with about 20 IP phones and 20 computers. Right now they are using a regular Linksys router which we use at homes. Their switch is also a very standard switch. Now they need to put there something better and VoIP compatible. What people use out there in serious and professional VoIP installations for medium size businesses? Is there

Hello- Has anyone had experience connecting to a Marconi switch (in the UK) using E1-PRI connections (TE410P)? In a new installation, my customer is getting yellow alarms on every channel about every 30 seconds. These alarms clear themselves immediately and then re-occur in another 30 seconds, ad infinitum. I used the same configuration that I've used successfully in the past while talking

Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router using MGCP IP protocol, instead of controlling it through an E1. Have anyone tried this configuration? How does MGCP works? I've tried to search for it on Google, but I only find the protocol specification for it. Is Asterisk fully capable of this? I can't find any documentatin covering the use of MGCP in Asterisk.

Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other

Ok I have googled and googled and combed through the wiki for an answer to this and have come up empty. What I'm finding is that when a user changes their VM password, it is saved somewhere like maybe the CSV database or something because when you log in, the new password works fine, but it's not saving to voicemail.conf. So new passwords are lost when asterisk is restarted and people

Hi, I'm looking for opinions on the "best value" router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source