similar to: System hardware requirements for *

Hi, I'm looking for a multi-line sip/ip phone that can answer multiple incoming paths. IE a secretary sits at a front desk and can answer multiple incoming lines/DID's. Is there something like this? Thanks, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041202/5b685d69/attachment.htm

Ok, so I'm setting up my small office. I have my asterisk machine setup and I have 3 sip phones connected as my stations and a 4 port FXO card ready to go(planning on only using 2 lines currently). What should I now order from my telco(sbc in this case) Everytime I call, they want to sell me this expensive $50 package that bundles everything and that's for a single line. Is there a

Paul, Ah, I see. Our echo is largly under control now. It took me a while to figure out the gains and get them tuned, and now the echo only leaves very small artifacts. Nonetheless, this still provokes the odd complaint here and there. We use VOIP for outgoing calls when our POTS lines are congested, and we find zero echo during those calls. Therefore, I assume that our handsets (Cisco

Brent- There is no good way to say what changing the hardware and PSTN hookup will probably do for the echo problems. I'm not sure if you mentioned (lost in the past history of your post now) what sort of hardware you're using for PSTN connection now- TDMs, X100s, ATA's, etc- but that could also be a potential cause. I've heard tell of aftermarket X100s and certain ATA's

I have a Dell PowerEdge 4300 server with dual Zenon 550 MHz processors, SCSI controller with four 9MB drives and 1 GB of RAM. I want to develop an asterisk pbx with 4 POTS lines in and 16 analog extensions (no VOIP). I also will install a sound card for an intercom. Is this hardware sufficient if using a Digium TDM2400P? -- Thanks, David Little M&M Technology, Inc. david at

Out internet connection was out this morning. It seems that the SIP extensions on our LAN were affected. Behavior like: Call comes in over POTS to a TDM400P, there is a delay then before the Cisco 79[46]0's start to ring. If we were lucky enough to get a call through, then we could not transfer the call, or place the call on hold, or park the call. Outbound calls seemed to have a delay

So... I upgraded via RPMs (actually, I uninstalled 2.2.7, and then installed 2.2.8 because of the dependencies... Is there a better method to upgrading?) this past weekend. Monday, end of the day, the entire machine crashed. No unusual circumstances, it sits there acting as a PDC on a Mandrake 7.2 Pentium 200 with 40MB of ram with not much else going on. Suddenly the whole machine locks up,

Hello List: Am a newbie planning to setup Asterisk for the first time. Planning to use PIII 550 Mhz with 256 or 512 MB Ram & 80 GB HDD. Please advise if the above configuration is ok for the application below...if not, please tell me what I need to have a comfortable system. I plan to install minimum CentOS 4.2 Server required with iptables, Samhain & Asterisk & associated programs.

Looking for a mentor... Having some issues with Asterisk 1.4.22 install. I am new to both Linux and Asterisk, however have 20+ years programming experience. First off I hate asking questions I could answer myself. I have and am reading The Asterisk manual, 2nd edition. I have successfully installed CentOS 5.2 and used yum to get a C compiler, current speed bump is with ./configure bash:

There is a 'feeling' on the opencores list that an audio decoder is desired. (MP3 gets mentioned occasionally, but has its patent problems etc...) http://www.opencores.org/ no audio projects are underway yet and I think a vorbis decoder will take a while but even if only a start is made then others may add to it. For their one year anniversary they will actually be doing a chip run and

Hi Lists, I have one box with two FXO and two FXS ports, it is running asterisk inside. I have one sinle POTS line connected to the one FXO and two phone sets connected to the FXS port. Extension 6003 is asigned to one fxs and 6004 is asigned to another fxs, the two extensions can call each other, they can both make/receive PSTN call, but they can't make PSTN call simultaneously. Is it

i have 6 pots lines coming in from the outside world (but we are reducing to 4) all the lines have the same phone number. i have 40 analog telephones that need to be connected to them. one way (i think) i could do this is to have a asterisk box with a tdm400p with 4 fxo's connected to the pots, also in the asterisk box would be a pair of t100p each connected to a tsu-600, the tsu-600 would

I am having a problem with Red Alarms on X100P cards. The most frustrating thing is I can not duplicate the alarms, therefore am not sure how to solve it. I have read after searching posts and the web that you can try different signaling methods which may help alleviate the problem. There is fxsks (Which I am currently using), fxsgs, and fxsls While reading the Digium site about these different

Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a "call forward" option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them "try out" our dialtone

We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This should get rid of static on the line (at least any static generated by our half of the circuit), right? I am a total virgin to ISDN. I understand that we need two BRI circuits to provide four voice channels, and that the hardware to speak to the BRI circuits can be passive or active, with the active type being much more

Hi All, Got a strange (at least IMHO) issue that doesn't make much sense to me. Basic configuration is two sites with a site-to-site (aka router-to-router) VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks, and the only VoIP is internal - all of our outward telecom is on POTS or Centrex-enabled POTS lines. Site 1 has a Dell PowerEdge 1950 with Asterisk built

Hi all, Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones. What would you recommend to use as the 8 FXS switch? I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it is the best solution. Does anyone have a better solution to build this system? If an analog

Hi all, I''ve been following the puppet project for a while now and I think it''s starting to become mature enough to be used in a pilot project at my corp. I have a few questions: 1. What is the current largest known number of Puppet managed hosts? 2. What are the average network demands on a single puppet master? 3. Has Oracle ever been managed by Puppet? 3a. Has Oracle ever

All, Thanks for taking time to read. I am wanting someone to tell me that I have a clue here and that the following proposed setup would (possibly) work. Currently existing are 4 pots lines with qty. 10 AT&T 954/854 phones. Would like to convert to asterisk and at the same time do a mild feature upgrade. Proposing to get: PIII 800MHz or better >=256MB RAM

I have two servers each with one TDM card from Digium and three POTS lines going into each server (each POTS line is one individual number). and I want to have all incoming calls into server B be directed to the IVR on server A and to be able to fail over to Server B in case Server A has an issue (it can be a manual failover and I actually prefer it to be a manual process). I've