similar to: VoIP dialtone?

Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of "power hubs", but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be

Hi all, This is a NEWBIE question, so all you experienced types that are tired of stupid questions can move on... I've pretty much given up trying to do my entire phone system over IP (including local service), so I have to select and provision my local CO lines. I need about 10-12 lines which can be POTS lines, of course. But, I thought, why not get something digital and expandable

Okay, I am facing a move in two months to newly renovated space. I have to decide *this week* between: A) Pull LAN and phone cables, prepare to move and expand our "traditional" PBX (Panasonic KX-TD1232 and VPS200). or B) Pull only LAN cables, go VoIP, use Asterisk as PBX. It is *not* an option to purchase a VoIP system package from Cisco, 3com, etc. Installers are getting an

Hi all, To solve my need for dial in modems, I've hit upon an idea: buy a used T1 "analog" modem bank like a Lucent Portmaster that takes in a T1 and provides several 56K modems. This is overkill for a lightly used dial in service, but the prices of these boxes is so cheap (~$300) with ISPs going away from dial in service that it makes sense. This is how I imagine it would work:

Hi all, I like to have a dial in modem on my toll free number so that when I or my employees travel, they can always get in for net access to read email if no better method is available. Right now, my Panasonic KX-TD1232D PBX receives the call on a POTS line and routes it to an analog modem. The speeds achieved are around 24 kbps due to the digitization of the KX-TD1232D introducing

Hello, I ftp'ed the binary RPM for samba from us6.samba.org, and when I tried to verify it, rpm said that MD5 was NOT OK. Has anyone else reported this behavior? Thanks, Hiten Sonpal sonpal@ciholas.com Engineer, CIHOLAS Enterprises Voice: 812 476 2721 x 102 2626 Kotter Ave, Unit #D Fax: 812 476 2881 Evansville, IN 47715 http://www.ciholas.com

Hi all, I would be most grateful if someone would review my plans for my new phone system and comment on areas of expected trouble and advice on what to do better. Instead of moving our Panasonic KX-TD1232/TVS200 system (ugh...) to our new location, we've decided to jump into IP telephony with *. But we are new to * (but not Linux), so we're trying to learn as much as we can before we

I have a dilemma. I am trying to setup an asterisk setup for about 10 people, but with the ability to expand to 100s. We are looking into hardphones to use for our systems. Here are the requirments I am looking for: SIP Not POE Full duplex Speakerphone $100-$300 We would like to have the cordless and corded option if possible (at least for 3 of the 10 people). So here are the

I've been looking at the two phones Aastra 9112i and 9133i for our offices now. I was just curious if anyone could tell me about the ring tones. Specifically: 1) How many ring tones do they come with? 2) Can you upload your own ring tones? 3) If so, what format and how? (wav, mp3, gsm, etc. and TFTP, web management, flash, etc.) Thanks for any help. -- Tim DeBaillie 812-476-2721

Ernest, I hadn't thought of doing that, though having that added protection would be nice. However, what I'm trying to do it have an incoming call at my home number follow me to my cell phone for selected numbers -- Since I already have three way calling, I'd like get Asterisk to essentially three way my cell phone into the call (or my office number, etc.) I understand the

I'm running into some errors I can't figure out. Situation: I have a Sparc IPX (2.6 Solaris), a Mac OS X box and an Ultra Sparc 1 running Solaris 9. I am trying to rsync a web site between the IPX and Ultra 1. All boxes are running 2.5.5 rsync built from source. Problem: Can't get a a sync between the IPX -> Ultra. IPX = kashmir Ultra = vortac Mac = archer Assumptions: a).

I have 6 domain controllers that were running samba-tng using openldap, Six months ago I converted one of the controllers to Samba3.0.20 and exported my ldap information to be compatible with Samba, every works fine. Specifically I could change passwords at the local windows machine. Two weeks ago I converted my other 5 controllers to Samba3.0.21b everything works except changing passwords at

I'm testing out my SNOM 200 phone by trying to call out through NuFone. When I do so, I don't hear an echo at all (in fact I can't hear myself through the phone) but the callee can hear an echo when she speaks. NuFone tells me their network is totally digital and so can't be involved in an echo. This is all well and good, but the echo is still there. Any suggestions? As a

I am using LDAP as my backend. I have 6 PDC's running Samba 3.0.21b, each domain has a different SID. I store all user,groups,and machine accounts in one LDAP database. So that when I create a user once, all domains can see the user. This keeps me from having to create a user account on each domain for cross domain file sharing. The behavior for Domains running Samba 3.0.21b is a follows.

We are trying to implement "area-code dialing" in our asterisk PBX. Basically: we will have a number of customers, who may be in different area codes, that want to direct-dial each other's extensions. We want this to work like a "real" centrex, in that seven-digit numbers should try (1) "local" VoIP extensions, and then (2) "local" PSTN numbers.

I am having some oddness with the 11/11/2003 CVS of *. Specifically, outgoing audio to NuFone doesn't seem to be transmitted (I can hear the other side just fine). My firewall is set to allow all outgoing traffic, and the IAX2 connection is definitely established correctly. Also, I can watch UDP traffic going by on the firewall so I know that * is transmitting. This happens with X-Ten on

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

Hi, Does asterisk support any kind of voice encryption? Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031224/f06c2d25/attachment.htm

Hi, I have setup Icecast2 and edcast (standalone) on an XPSP3 machine. The basic setup works. I can speak into a microphone and hear my voice from another PC on a LAN. Now I want to "stream a file" such as an mp3 podcast I made or just a song (mp3, acc etc.) or play a CD and don't know what to even search for in the mail archives, forum or Google to get help. I can't

Hello, We have an otherwise working Samba PDC+LDAP setup but for one minor detail. The shadowLastChange is not updated when the user sets his password from a Windows workstation (all the necessary Samba*-fields as well as userPassword is updated in LDAP). We've enabled 'ldap passwd sync = Yes' in smb.conf. slapd.conf contains the appropriate access rule for the said field. The