Dear all

i’m planning a migration to asterisk for a high volume IVR service
(from 1000 to 1500 concurrent call)

The IVR service is based only on DTMF tones so the features required is

- play feature
- dtmf detection

Asterisk will receive calls via VOIP (SIP with g711 codec)

The IVR service wil be a static service based on Asterisk dialplan with some prompt (from 0 to 5, play of files in the same codec of the received call) and some dtmf detections.

How many simultaneous call can i handle per server? each server will have:

4 core 3.0 Ghz
4 GB of RAM

I need an aproximate sizing:

0-100 calls per server ?
100-200 calls per server ?
200-300 calls per server ?
300-400 calls per server?
400-500 calls per server?

Thanks to all in advance

This is something I’ve seen with some key systems and PBXs. When the phones are on-hook, they can play music throughout the office instead of having an overhead speaker system do it. Never heard of it being done with VoIP, but figured I’d ask if anyone else has. I don’t see any way to do this on any phones I’ve looked at.

Hi,

I’d like to replace my current VOIP provider with an Asterisk based solution. I have some ideas I want to run by the list to see if they are possible, and get answers to a couple questions.

I want to setup two Asterisk servers that are linked to each other:
- The first server would be my “external” (public) server and would live in a real data center. The second server would be my “internal”
(private) server and would live in my house.
- The external server would receive all incoming calls and handle the voice mail stuff.
- The internal server would run all the phones in my house (VOIP or Analog-via-FXS). All outgoing calls would be routed out through the external server.

I also want to add the following additional functionality:
- If the external server looses connectivity to the internal server while a call is in progress, the external server should place the call on hold while it tries to reach us via our cell phones. A message should be played informing the remote party that the connection had been lost and it is trying to re-establish it now. If it can’t reach us, it should inform the remote party that the connection could not be re-established and allow the remote party to leave some closing remarks on the voice mail system.

- If a call comes in and no one is at home to take the call (or if all lines at home are busy), it should ring all of our cell phones and whoever answers the call first gets the call. If no one answers the call via the cell phones after 3 rings, it should route the call to the voice mail system. I say 3 rings on the cell phone because I do not want the cell phone voice mail to take the call.

- I also would like the system to automatically route all calls directly to voice mail depending on the time of day (say 10PM to 8AM).
I would like specify in a “white list” specific phone numbers that are allowed to ring through regardless of time of day (i.e. her parents, my parents).

- I would like the VOIP phones to turn on the voice mail waiting indicator light if the external server has new voice messages.


Is all of this possible? If not, which part’s are not (and how much work do you think would be needed to make those parts work)?

What is the best way for me to setup Fax Capability with VOIP only.

I have a Asterisk Server hosted on the internet without a modem. I’m using Flowroute, which is working awesome, for VOIP calls.

I only have a SIP Phone at home and two Printer/Scanner/Fax Printers.

I’m not sure which Fax Addons or Extensions I should use for Asterisk.
I’d like it to Self Detect on any line.

I also am not sure what or how I can connect a Network Only Printer/Scanner/Fax Machine at home to it. It has a Telephone Jack but I’m only using VOIP.

I’m pretty advanced with Asterisk now and can figure things out..I would just like some advice and direction before I get started.

Oh, one more thing. Is there any way to Route the Faxes to different folders (extensions) because I have End Users with Phone + Extensions when you call in.

I’m running Asterisk 1.8.11.1 and am connected to the nexVortex trunking service over a DSL line solely dedicated to VOIP usage. For both incoming and outgoing faxes, I’m getting a failure rate of just over 25%, and over a handful of reasons.

Is it natural to have this many problems on a completely digital configuration? I’m trying to cut our analog phone line (because it’s so expensive), but some fax machines just don’t seem to ever accept a fax. Many of the failures are on the same numbers, forcing me to fall back to an old analog fax machine just to make sure it actually gets through.

Has anyone else had any similar experiences, or is this indicative of a failure in the setup on my end (or even the trunking service)?

Brett Lehrer