search for: squiz

...transcoding to g729 across the IAX2 link. Thanks, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net ..... > > Open Source - Own It - Squiz.net ...... />

...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: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ .....>> Open Source - Own it - Squiz.net ...../>

Is there any way to customize VoiceMail ? I would like to customize the message played to callers sent to the voicemail becouse the extension is busy or otherwise unavailable. Is it a way to record a welcome message and use it ? thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it

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

...ut I'm not actually sure what to look for! Thanks, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ .....>> Open Source - Own it - Squiz.net ...../>

I have used a channel bank to mux together BRI's and partial PRI's to a single PRI to avoid the goofyness of chan_capi. The focus of Asterisk support is geared to PRI's, it seems. -----Original Message----- From: Avi Miller [mailto:avi.miller@squiz.net] Sent: Thursday, June 15, 2006 7:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] what are the elements of a good asterisk setup? Tyler Retzlaff wrote: > to use Active card(s) and I need to accommodate 2 x BRI TE/PTP > services. I...

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

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,

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

...seem to be any of these available at the moment. Thanks, Avi -- National Manager - Special Projects < Sydney . Melbourne . Canberra . Hobart . London /> Walter Turnbull Bldg T: +61 (0) 2 6233 0607 44 Sydney Ave, F: +61 (0) 2 6233 0696 Forrest, W: http://www.squiz.net/ ACT 2603 .....>> Open Source - Own it - Squiz.net ...../>

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

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

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

...way to manage the directory from a central location. Thanks, Avi -- National Manager - Special Projects < Melbourne / Sydney / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ .....>> Open Source - Own it - Squiz.net ...../>

*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?