similar to: Cisco 30VIP Phone

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

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

I am looking to replace all of the old "Bell" (POTS) phones in my home and office with IP phones. As you can imagine I don't have a huge budget to work with but I want phones that will provide acceptable voice quality and durability. There are basically three categories as I see it 1. satellite phones (low cost, low function) 2. primary domestic phone (good quality, POE capable,

Dear All, I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk??

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

While I'm frantically scouring this list, does anyone have any information about getting BLA (busy line appearance) working on Cisco 7960? The last I heard was that this was unsupported in Cisco's SIP firmware

Can someone provide an example of how to use HINT priority with Cisco 7960/SIP phones? I don't fully understand what exactly the hint does, but I believe it mimics a legacy PBX's bridge-appearance function. Is this correct?

I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones

Hi all, I'm new to Astrisk so bear with me. I have just installed AsteriskNOW and am quite familiar with RH Linux. I have configured it and am using Xlite to connect and learn to move around the conf files. I have a problem, however. The client connects and dials ok, but there is no audio. In searching the archives I found discussion of this issue primarily centered on NAT issues. This is

Hello, In voicemail.conf, it's possible to edit the voicemail message, but when I define a pager email address, I get the message from "Asterisk PBX", and the content is fixed by the system. Is there a way to manipulate this message, as well? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, Does anyone have any experience getting a 7936 to work with Asterisk? Do you need to use SCCP or is there a SIP image for the phone? I have a few 7960G's and they are working with SIP, just curious if the config of the conference phone is the same and if anybody has any good setup links. Thanks! NB -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there anyway to force an autopickup on a cisco 7940 / 60 from the dialplan ? My problem is that I am originating a call from the AMI, with the internal user being called first, and then connecting to external user. However, sometimes the internal user doesn't pick up the phone, so the call is never placed. I need to know the results of the call so I need to be able to either a) get

I'm wondering if anyone else has experienced this. Up until a few days ago, when accessing the CLI from my terminal program (Private Shell), the output was in color. I haven't upgraded, rebuilt, or to my knowledge, changed anything in Asterisk that would change this. My terminal settings were the same as well. I have two computers that I access the CLI regularly on, and neither show