similar to: Newbie Question

Hi all, I have just successfully configured a Cisco 30VIP to work with my Asterisk server. I have seven of these phones new and would like to deploy them. I am wondering if anyone has this phone deployed with Asterisk and can suggest configuration of the various buttons, etc. (Bare with me as I am new to Asterisk.) Thanks, Chris

Hi all, Is there a "speed dial" type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar

Hi all, I'm working on a script to check the status of streams from multiple streaming sources (ie. icecast, shoutcast, wms, etc.). With Icecast I am looking at the HTTP header returned by the server when I do a GET on the mountpoint I want the status of. It appears that a HTTP/1.0 200 indicates that the mountpoint is up while a HTTP/1.0 404 indicates that it is down. Two questions: 1. Is

What are best practices for allowing connection by remote SIP extensions over the internet? I'm thinking of putting the SIP inside a VPN connection. Kind Regards, Chris

Hi all, I'm new to rsync and have a problem I cannot resolve even after plowing through the lists. I have an rsync server setup with an rsync client. I am trying to backup a directory structure that is over four levels deep. Rsync does great until it hits that fourth level. Then it errors out for each file in that fourth level similar to this: rsync: recv_generator: mkdir

During the course of a conversation with an member of the IT group who handles the E911 center for our county, I learned that all of the county's E911 is voip based. This got me to wondering why we could not just configure up a SIP or some such trunk directly to the E911 center to handle our emergency traffic. The county seems interested in exploring the possibility. So I'm wondering if

Samba: 3.6.6 PDC Client: XP Pro SP3 Background: 1. Started with a clean installation of XP Pro SP3 2. Joined the client to the samba domain 3. Logged in as user 'root' the first time after the join. 4. Added "Domain Users" group to the "Local Admin" group on the client (forget about the security implications for the moment) Now when a user (any user) logs on to the

I'm starting a new thread because the issue now is different from the one I was originally experiencing. Here are two level 10 debug logs one from each of two servers: http://w4fbc.org/samba/DC.log http://w4fbc.org/samba/FS.log Each has a test share setup like this: chgrp -R staff-faculty /test chmod 0770 /test chmod g+s /test setfacl -m g::rwx /test [test] path = /test read only = no

I am running Samba 3.6.6 on a Ubuntu 12.10 Samba domain member server. Users are authenticated against a Samba DC running 3.6.9 over an LDAP backend. I have a share configured as show below. Members of the 'staff-faculty' group can browse the share, but cannot write files to any subdir for which they are not the owner. It appears that the only reason they can read/traverse is because of

Hi, I'm running Samba 3.6.6 on Ubuntu Quantal. I have a need to manually assign some of the UID mapping on a samba domain member file server. I have used tdbtool to add the correct mapping record to winbindd_idmap.tdb. However, I am at a loss as to how to force that change to "propagate" so as to show in the permissions structure of the file system and in the output of such

Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty

I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

I've just seen this post: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is

I can't get my Samba PDC (FreeBSD 7,0-BETA3) changing UNIX passwords from Windows clients (Ctrl-Alt-Del). I now have the password chat debug active and I have loglevel 100. I am not certain about the syntax in the password chat. But if I from a console try to change the password of a given user (here testuser1), I see these lines: mflserver3# /usr/bin/passwd testuser1 Changing local password

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the Xlite ends up using the ulaw codec as

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.