similar to: Residential router w/ QoS support?

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

Hello all, Last couple weeks we had a lot of business discussions on mailing list, however some people don't like it, some people don't needed it, etc. I had couple discussions with Asterisk community members, who is interested to have business discussions about Asterisk, including but not limited to : business implementations, reselling , Asterisk commercial packages, IP phones,

I have a question on SIP devices that are setup and working but you change the login name and contents to them why does asterisk need to be shut down and restarted for them to work? I have reloaded extensions and done a reload command. But the updated sip phones do not work until I shut down and restart asterisk. Is there any other way to update them without restarting the system? Since the

hello, Is there a way to transfer the call via gnophone, without calling other user and pressing conf on both calls, it seems that all traffic is still going through the gnophone, not that optimal i guess. thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

Has anyone had any success using more than one or two X100P cards? I have 4 in a system, and channels 2 3 and 4 all seem to work just fine. Channel 1 however is acting up. I get random red alarms, disconnects, etc. I have checked the /proc/interrupts and everything is sitting on it's own IRQ. Also checked memory addresses and everything looks good there. Not looking for anything specific,

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

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

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

I see in the changelog that ENUM support has been added.. anyone know what this is? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hello Folks, I noticed that all of the IaxClient based softphones with exception of Firefly only seem to have support for GSM but what about iLBC? The quality is excellent with iLBC even on a dialup connection! Meanwhile while the audio on GSM often sounds scratchy. Is anyone looking to implement iLBC in an IaxClient based softphone soon? Errol Biz4Web Solutions Limited

> > --__--__-- > > Message: 1 > From: Terence Parker <terence@parker.com.hk> > Date: Fri, 9 Jan 2004 11:25:23 +0800 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Problem registering FWD > Reply-To: asterisk-users@lists.digium.com > > > --Apple-Mail-1-822243116 > Content-Transfer-Encoding: 7bit > Content-Type: text/plain; >

> > JR Richardson wrote: > > > I?ve got Asterisk up and running on a VIA mini-itx M10K with a TDM-400 > > card but I?m having a terrible time with the sound setup. I tried RH9 > > and Debian distributions, with 2.4.18, 20, 22, 24 and 26 kernels. I?ve > > used the VIA drivers, ALSA drivers, OSS drivers and Kernel drivers > > with varying results. I can

I've been looking for a good iax or sip <==> ptsn provider. Someone with very low cost usa calling and can offer incoming ptsn connections in most markets. The only decent providers I could find were iconnecthere and nufone. Has anyone found someone that really stood out? Matt Hohman New Heights Church http://www.newheights.org 7913 NE 58th Ave. Vancouver, WA 98665 Office:

Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. bkw

Hi, I buyed a wildcard e100p and I installed that but when I load asterisk I have this error: Apr 29 18:59:25 WARNING[8192]: chan_zap.c:690 zt_open: Unable to open '/dev/zap/channel': No such device Apr 29 18:59:25 ERROR[8192]: chan_zap.c:5403 mkintf: Unable to open channel 1: No such device here = 0, tmp->channel = 1, channel = 1 Apr 29 18:59:25 ERROR[8192]: chan_zap.c:7526