similar to: RE: Asterisk-Users digest, Vol 1 #3387 - 9 msgs

Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com

community there is a real possibility this may come off so if you have an interest in this space and want to contribute to the discussion then this is your opportunity to do so. =20 I look forward to all opnions on this topic. =20 The slide deck for the agenda of this call is located here http://voipusersconference.org/2008-05-09-Slides=20 Cheers, Dean=20 ________________________________

I am out of the office until Tuesday Aug 6th. If you need immediate attention please contact Don Jemtrud at don@magnumlog.com or call 297-6205. >>> "samba@lists.samba.org" 08/04/02 08:26 >>> Send samba mailing list submissions to samba@lists.samba.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.samba.org/mailman/listinfo/samba or, via

1.) Edited my SMB.conf inside the /etc/samba directory - I changed the = default workgroup name to match that of my Windows 2000 SP2 machine. I also edited the Public share to point to /home/markdog/apps = directory with open permissions for viewing. 2.) Manually started the SMBD and NMBD daemons- these are the copied = lines from a terminal prompt: [root@localhost /root]# smbd -D smbd -D

C:\>net view \\mummy System error 64 has occurred. The specified network name is no longer available. C:\>ping mummy Pinging mummy [192.168.2.2] with 32 bytes of data: Reply from 192.168.2.2: bytes=3D32 time=3D63ms TTL=3D126 Reply from 192.168.2.2: bytes=3D32 time=3D63ms TTL=3D126 Any ideas? Thanks. ------=_NextPart_000_004B_01C281B0.68A07A90 Content-Type: text/html;

through the guest account. This is my smb.conf on the linux side: ; Copyright (c) 1999 SuSE GmbH Nuernberg, Germany. ; [global] workgroup =3D SYAC guest account =3D winuser ; guest account =3D nobody keep alive =3D 30 os level =3D 2 kernel oplocks =3D false ; security =3D user ; Uncomment the following, if you want to use an existing ; NT-Server to authenticate users, but

found response from asterisk. =20 =20 On asterisk's log I see messages like: "Looking for conference on conference-context (domain serverIP)" =20 And: "Call from 'conference' to extension 'conference' rejected because extension not found." =20 =20 Does anyone have an ideia of why I'm getting that message? =20 Why does asterisk seem to be using domain

i click on a directory on the sco box that contains more than 200 files = in it, i either get a truncated list of the files in the directory or i = get a windows message box saying that the directory is not accessible, the specified server = cannot perform the requested operation. Performing the same operation from a windows NT 4, or windows98 computer = on the same network works great. Do I

---------- In?cio da mensagem original ----------- De: asterisk-users-admin@lists.digium.com Para: asterisk-users@lists.digium.com Cc: Data: Mon, 12 Jul 2004 11:48:05 -0500 Assunto: Asterisk-Users digest, Vol 1 #4502 - 11 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide

- the account is enabled until password expiry and I have no kind of = notification about the expiration of password.... - then a nice day the password really expires and, instead of been = forced to change it, the user is simply LOCKED OUT (account disabled) Note that in UNIX the user is still active as I've set a long Interval = beetwen pwd expiry and account locking ! The only

network outages and recent tests have shown that works well, albeit the switch takes about 20 minutes to propagate the dns updates but otherwise flawless. =20 It's embarrassing and I'm losing credibility when clients are asking if I'm still in business as the phone has dropped way to often in the past few month. Interesting enough all outages to date have been Fridays or Mondays. =20

using Asterisk. =20 Is this all you want Asterisk to do? (eg as an application service rather than provide telephony for the rest of the library as well), or are you looking to have it replace your existing telephony equipment? =20 As a suggestion if you google Trixbox and Nerd Vittles you will find a fairly detailed explanation of how to set your Trixbox server (a version of Asterisk) up to

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

using Asterisk. =20 Is this all you want Asterisk to do? (eg as an application service rather than provide telephony for the rest of the library as well), or are you looking to have it replace your existing telephony equipment? =20 As a suggestion if you google Trixbox and Nerd Vittles you will find a fairly detailed explanation of how to set your Trixbox server (a version of Asterisk) up to

And I have verified I am typing in the correct password. So it looks like there is some problem with encryption or something. smbd and nbmd are version 2.9999 (Debian packages) Thanks. --James Lamanna ------=_NextPart_000_006A_01C2864A.76DEA5C0 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML

i can register to asterisk with msn, place calls, etc. However, if I = unplug the ethernet cable and get a dialup connection I get the = following error when i try to place a call with MSN (i can register = withoug problems); chan_sip.c (process_sdp) No compatible codec! I have disallow gsm and allow ulaw on my sip.conf Any help would be greatly appreciated. Dan

When I first configure the client, in the Received Items, it shows all = items correctly, but then, when starting to donwload, they simply = disappear. All other folders are OK. Problem is just in inbox. I thing (although my tests were not exaustive) that deleting the .imap = folder solves the problem, but I just can't understand why, because it = is recreated again... Any help? Thank you,

then the RTP must go through Asterisk since we need to know if any DTMF = was pressed. If both users and the server are not behind NAT and we set up for the = RPT to go direct if I am using INFO for DTMF, when using the t or T = option does the RTP go direct ? I am asking since from what I understand out of band DTMF was created so = that a leg in the middle can get the DTMF with out having to

running in root. Mine is running as asterisk. I found one post that says to run at boot: #!/bin/bash=20 /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP = --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 10000:20000 -j = DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP = --set-dscp-class ef Does this

might be your best bet to get the information you want. I'd look at voip-info.org for information. _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier Sent: Wednesday, September 16, 2009 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to list ongoing calls