similar to: how to customize voicemail

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 to set up an Asterisk with an 4-port BRI card. Hylafax should send and receive fax. Will this work reliable? Any recommandations for an 4-port BRI card? Other alternatives except analog fax units? thanks for your help best regards Thomas

Hi Guys Just wondering if you can help me and give me some pointers on what could be wrong in my setup. Everything seems to be installed. I have Asterisk 1.2.12.1 FreePBX 1.2.3 IAXModem 0.1.15 Chan_capi 0.7.0 The IAXmodem registers, faxgetty is starting in /var/log/messages but it just doesn't seem to pickup the call. In Chan_capi it says fax detected but no fax

I thought I could try to post this question here since I have at times found excellent support here on A@H issues as well as pure/straight digium/asterisk issues. I am looking to finally build a permanent asterisk server and would prefer, if at all possible, to stick with the "new" A@H called trixbox. Does anyone know if the CentOS behind trixbox is a relatively complete CentOS system? I

Can not check voice mail-box externally. I'm trying to log-in externally (from PSTN line) to check my "voice-mail" so I created context to authenticate log-in ... exten => 7,4,Authenticate(01894546) exten => 7,5,DISA(4789|disa-access) Authentication works OK, I get inside dial none enter my mailbox extension but it doesn't accept my mailbox password even though it is 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

Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks,

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Good Morning, I've been trying to set up asterisk@home and thing are going pretty well. I do have a question: When I *98 into voice mail I hear a message that says "Asterisk mail" then short pause then the word "mailbox" then a very long pause, then a request for a password. I believe some other audio should occur in the long pause between "mailbox" and

Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being acceptable in US, but in UK its a right pain). As soon as the 10th digit is entered, it starts to dial and the number is invalid. If the phone is left

Hi I make a Digital Receptionist ,but how can I attach it to an extension Help regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic

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

We are moving our asterisk 1.0 system to a new Asterisk @Home system (2.8) and I am the one in charge of doing it. I have run into a snag, though, on meetme conferences and with the transfer key. Regarding the transfer, it appears that both directions of all calls can transfer by pressing the # key. I do not like that ability. I would like to change it by doing 2 things: 1. Make the transfer

*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

-----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

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, 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:

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

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

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