similar to: Digital Receptionist

Good day! so I've inherited an Asterisk PBX setup and it's busting my ass. Since the setup wasn't document and is not running on our standard operating environment I have to say it is an ad-hoc / experiment which has become critical infrastructure for the company and is now considered a risk. Because of this there is a need to rebuild and (this time) properly document the

Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed

Hey guys, Is anyone aware of a single PRI (E1) card that has onboard hardware echo cancellation? I can only seem to find 4-port cards (Digium and Sangoma) and I really don't need the other three ports. Thanks, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F:

Hi I've got a dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out using the acopy2 test utility. I'm having trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where to look ? I can attach conf files etc. if needed. Asterisk says it has 30 capi channels available,

Hey guys This is probably for Armin, but I thought maybe someone else might have something I could try. I'm having a problem with one of my Eicon Diva V-4BRI cards and I'm trying to work out if its a driver configuration error, card failure or telco problem: I have an Asterisk box running -- the Eicon drivers see all four ports on my card, capi.conf is configured with all four ports

I am intending to rebuild our asterisk server on a Dell 2800 with Dual CPUs. I am currently using Centos release 3.5 (Final) with the 2.4.21-32.0.1.ELsmp Kernel on a Dell 1750. I have a TE410P (Quad PRI) card and a TDM card. I remember that there were issues with the 2.6 Kernel, but Iam sure that it was with the older asterisk builds. Also, the new server has the X64 intel CPUs, so I would like

Hi! I'm always getting this error when echo cancellation should start. == ISDN3: Setting up echo canceller (PLCI=0x203, function=1, options=4, tail=64) == ISDN3: Setting up DTMF detector (PLCI=0x203, flag=1) -- ISDN3: Error setting up echo canceller (PLCI=0x203) Apr 3 10:49:30 WARNING[17266]: chan_capi.c:3334 show_capi_conf_error: ISDN3: conf_error 0x300b PLCI=0x203

Using FTP to configure 501. Gets past "Running... App = sip.ld" and: Welcome! Processing configuration... "This may take a few seconds." Then it displays: Config file error Error is 0x4020 and reboots continuously, repeating the above. Anyone seen this before? Is this a corrupt *.ld file? An FTP error? (PASV)? A missing SETUP setting? Thanks! -- Jim Rice by Design

*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * *0 Dial option is tTwWr I tried to call from 601 to 615 601 keys in *0

Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I'm in charge of two Asterisk servers. One of these servers is running an Asterisk@Home version under CentOS, the other one is running Debian Sarge. The Debian server has been installed by me using this howto: http://www.astpp.org/index.php?n=ASTPP.DebianSarge-AMP On both servers, the AMP version is 1.10.010. However, the

Are there any step by step instrunctions on how to install drivers and I guess bristuff for this card? Just need to use it to handle voice on 2 BRI circuits (UK) then utilise with Asterisk and some Digium cards handling POTS phones (and some VoIP out the back). It's the EICON card stuff and how to make it all work I'm finding confusing? Steve -- NetTek Ltd UK mob +44-(0)7775 755503

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

has anyone managed to get these three beasties to work together ? we're using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft netmeeting default from windows xp. the symptoms are that calls from a SIP client to NetMeeting rings on NetMeeting, but upon answering the call in NetMeeting, no audio is passed between the two. eventually, the call times out and hangs up. on a

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Hi! I'm here again with echo canceller problem... :-( I think I've done everything to enable echo canceller feature, but it still doesn't work... Can anybody tell me if there is some error or something missing in this configuration please? I'm using Eicon Diva Server 4Bri. http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm?techspec=1&regID=4999 Card

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?

Hi, I'm running a 1.1 version of Asterisk (a stable build from back in Oct-05) running on RedHat 9.0. Everything's been great but a couple of days ago, we all stopped receiving emails of our voicemail. There's been no changes to our configuration I bet I'm expereiencing a Linux problem rather than an Asterisk problem, but because I know only as much Linux as required to get

Hey gang, I'm hoping someone can help me out here. I've just noticed that on two of my five Asterisk boxes (CentOS 4.4, Asterisk 1.2.12.1), I'm getting the following translation cost for g729: asterisk*CLI> show translation Server 1: g729 - 26 25 25 25 25 24 26 - 53 36 Server 2: g729 - 66 65 65 65 65 64 69 -

Hi I have a TDM card with 4 lines on a hunt group coming in. I can answer the phones with exten => s,1,Answer() exten => s,n,Dial(Zap....&Zap....) ... The problem is I don't know how to find out what extension was originally dialed. And, trying to match on the extension always fails. E.g. exten => 1234567,1,Answer() # never gets here I thought I could get the extension on