similar to: Tutorial for SIP user

Hi all, I did some call using an asterisk 1.4 PBX and 2 softphone in a private network; call is up, but with bad quality. Someone knows how to debug this problem ? Thanks in advance for any help. -- Giancarlo Lombardo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091111/83d6d4ff/attachment.htm

Hi all, I just configured some user in sip.conf and extensions.conf; they works fine. Now I'm trying to do the same with Extensions feature of FreePBXAdministration, but I cannot see what I have done manualy. Is gui working with other an source, How can I access such data ? Thanks in advance for any suggestion. -- Giancarlo Lombardo -------------- next part -------------- An HTML attachment

I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091027/bb5e7b1a/attachment.htm

Dear all, I just installed asterixnow, but no graphical interface start automaticaly neither linux nor some other, just command line. Shall I do something or shall I install something more ? Many Thanks in advance for any help. Giancarlo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers -

Where is everyone located? I am in Washington DC. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/7c73847d/attachment.htm

Dear all, I'm setting up a connection via XLITE softphone and asterisk 1.4 but I get the error: *Registration error: 404 Not found* Here my configuration file of asterisk: *[root at dhcppc0 asterisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* *[root at

Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten

Thanks Tristan The patch is attached and applies to speexdsp's master. The warnings came from Wireshark's coverity scans. Dario. On Tue, Sep 20, 2016 at 10:56 PM, Tristan Matthews <tmatth at videolan.org> wrote: > Hi Dario, > > On Wed, Aug 10, 2016 at 6:07 AM, Dario Lombardo <lomato at gmail.com> wrote: > > Hello devels > > I'm a core developer of the

Hi all, As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00 pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. Any idea or solution. Regards Farooq --

Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable.

Hi, Asterisk has a lot of customizable voice prompt in /var/lib/asterisk/sound but i want to change a very well known voice message which occurs when we try to dail a number against dial plan "beep beep beep The person you are calling is unavaiable, please try again." I thought it would be availabe in the sound directory of asterisk but it is not there. When i dial such wrong number no

hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq --

Hello Everyone, I am receiving following error message will making Zaptel on Cent OS 5.2. make[1]: Entering directory `/usr/src/zaptel-1.4.12.1' echo "You do not appear to have the sources for the 2.6.18-92.el5 kernel installed." You do not appear to have the sources for the 2.6.18-92.el5 kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory

Hello, Not sure if this is implied in the documentation or something given: does the RoR framework require "every table must have an id" ? If this is the case, then what happened to the concept of normalization ? Futher, if the tables have already been build for an existing appication ( without ID column in everytable ) , will the script / generate work ? Thanks -- Posted via

Dear R-users, I couldn't find a function or some help in R-project web about the Generalized Singular Value Decomposition. In MatLab there is a simple function for this algebric issue (gsvd). Is there anything like that in R? And, if not, could you help me to apply this method in R? Thanks in advance, Giancarlo +++++ This mail has been sent through the MPI for Demographic Rese...{{dropped}}

Hello Everyone, I have write a script following script for nagios -- typeset -i CRITICAL; #Positional parameter CRITICAL=`echo $2`; ME=`basename $0`; #echo $CRITICAL if [[ "$2" == "" ]] then echo NO INPUT!!! Usage ./$ME -c N else typeset -i ASCALLS; ASCALLS=`asterisk -rx "core show channels" | grep active | grep call | awk '{print $1}'` #echo $ASCALLS;

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Dear R-users, I would like to know if there is any way to minimize || y - Xb||^2 under the constraint b'b=1. Thank you Giancarlo

Hello devels I'm a core developer of the wireshark project. In our audit process, a piece of code of speex (the copy we maintain inside wireshark) showed a flaw. I have a couple of patches I'd like to send you but I can't find details about the submission process. Can you help me? Thanks Dario. -- Naima is online. -------------- next part -------------- An HTML attachment was