similar to: IVR Menu and VoiceMail quality

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

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

Hi! We (an astronomical center) are building a small cluster for our number-crunching. The problem is that we have 12 mobos A7V600 by Asus (for athlons) - 1 server + 11 diskless nodes. And here comes the trouble... hardware: mobo: a7v600 net: 1 gigabit by 3com - 3c940, integrated in mobo the problem is: dhcp - works fine tftp - works fine pxelinux.0 - downloads fine (I hope) linux kernel -

I have a bridge that has gigabit interfaces. The machine in question has the fun job of being a Bridge, Firewall and SMB server. Both of the Gigabit interfaces are connected to workstations directly via Xover cable (well MDI-X to be exact). My question is, if I enable jumbo frames on the gigabit interfaces will that make any difference in overall transfer rate of the bridge? I was thinking it

How has experience in Asterisk voip provider? I am trying to setup a reliable Linux system with Asterisk for a voip provider. Therefore I got two more or like identical systems. System 1 AMD Atlhon XP 2200 Asus A7V600-X bios 1002 1Gb memory 333 Mhz Asus 7100 videocard 120GB harddisk System 2 AMD Atlhon XP 2200 Asus A7V600-X bios 1005 1Gb memory 400Mhz Geforce MX 4000 64MB 40

> -----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 >

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

CVS 7/16/04 (the latest one I have b4 today) seems to have this problem too... Anyone else having problems with the current CVS ignoring calls after about 5 minutes of being up? I've also noticed that no matter what I set default_expiry to in sip.conf, it starts at that number and then jumps to 44 seconds... not sure where it's getting the number 44 from, it seems to use that all the

I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them. Here is what I have in my sip.conf: [stanaphone] type=friend secret=pAsSwOrD ; skewed for this message. username=3475341914 host=sip.stanaphone.com

I have an idea, tell me if this wouldn't work... I know it's really ugly, but it might help some people until we can get round robin DNS checks for peers... Since * does not do GetHostByName() again until you reload your config, and BroadVoice and I'm sure other sip providers are using round robin DNS, why not create 2 [<your server here>-out] contexts in sip.conf, and then in

Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my

I originally sent this to samba-technical which as a typical first stpe for me was probably wrong.... ;-) Anyone here have any ideas? -----Original Message----- From: Petty, Robert [mailto:rpetty@denvernewspaperagency.com] Sent: Tuesday, September 23, 2003 10:07 AM To: samba-technical@lists.samba.org Subject: Samba 2.2.8a / Winbind and Domains I have searched the archived through google and

Hello, I seem to be having a strange problem here. First, I will describe my current computer system setup. Second, I will describe the problem. Third, I will pose several suggestions contributing to the problem. Fourth, I will ask you to investigate this for your self in search of quick resolution. Fifth, I will attach relavent configuration files. I have a production web server running

Hi all,? I have a CentOS 4.5 server running on a workstation mainboard (PCI Slots only).? We have now one 200 Gigs IDE disk dedicated for e-mail server storage.? We use Communigate Pro and the server has 45 Outlook clients with the MAPI connector (All mailboxes on the server).? When a user opens Outlook, a refresh of the local cache is performed for his data.? There is a big "Public"

Did my last post make sense? Is this a known issue with the bridge-nf code? Is there something I can do to help? Should I just shut up now? -Thanks in advance Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

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

Ok.. this is really wierd... I just cleaned up my dialplan a bit by adding some macros with a strange side effect... On my incoming context which has no macros in it, far end ringing used to work... now that I have macros defined, far end ringing has stopped working all together... The macros DO work, but when they transfer, the far end ringing sounds terrible and even skips a few rings... If I

Have anybody experience with the following error on a linux system. The system is for the rest running perfect without problems. The system was full installed with Suse 9.1 and updated. Following uname is my kernel 2.6.5-7.104-default Greatings Han Suse 9.1 professional AMD Atlhon XP 2200 Asus A7V600-X bios 1005 1Gb memory 400Mhz Geforce MX 4000 64MB 40 GB Harddisk

Hi Last week I installed Asterisk (release1) with digium t100p single span T1 (wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is currently the only user of the t1 line. All worked well for about a half a day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I consistently get multitudes of blue alarms on all b-channels followed by a loss of d-channel: Aug 31

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: