Asterisk as VoIP answering machine?

Asterisk Asterisk Support
1

Hi everyone!

I have few simple questions. Trying to configure * as pure answering machine, with VoIP (sip), on Linux.
Now questions…

Can i configure asterisk to pick up immediatelly,play short message, reads few digits (extension)
input from user , and then depending on input (defined in menu) play next message,and finaly hang up.

  • pure VoIP, no PSTN or digium cards, not even softphones.
    Do i need “softphone” “connected” to asterisk ? now i get only 'BUSY" tone, asterisk does not pick up. ?
    Do i really need audio card to playback .gsm files over ethernet - VoIP to caller?
    Can i do this on more lines ( more registered VoIP nr’s) at same time ?

I see that asterisk register my username:pass@provider.xxx/username at VoIP provider.
and for all nr’s i have. No complains there ( logs are now ok after i put ip of provider in /etc/hosts…)
but no answer. I read somevhere that even if configured, but clients(softphones) are not connected to asterisk, only busy tone will be showed?

Help anyone , please?

Greetings

mad071

2

This is very possible. I have done this, after a manner of speaking. I have a softphone account through Vonage that I use for people to call my asterisk server. In my case, I use it to prompt immediately for a conference call room number. In your case, you could simply have it prompt for whatever it is that you need and then allow the system to be navigated according to the recording. You configure all of this in your extensions.conf file. Please let me know what it is that you would like for it to do and I will be glad to help you write these instructions.

3

OK have you looked at VMWARE??

You can install VMware on the linux server and download
a prebuilt asterisk server with tons of stuff…Voice mail with web interface

You can grab the free VMware server at vmware.com/download/server/

and you can download a copy of TrixBox 1.1.1 with the VMwarez Guys / Nerdvittles tweaks pbx.mississippi.com/

you can be up and running in in a few hours (downloading will take longer than setup)

A little reading as the downlaoding is going on.

aussievoip.com.au/wiki/index.php?page=TB-Index

4

[quote=“mad071”]Hi everyone!

I have few simple questions. Trying to configure * as pure answering machine, with VoIP (sip), on Linux.
Now questions…

Can i configure asterisk to pick up immediatelly,play short message, reads few digits (extension)
input from user , and then depending on input (defined in menu) play next message,and finaly hang up.
[/quote]
Yes can be done :smile:

Can be done too !

You dont, but you will want some sort of phone for testing purposes.

Did you try not to register your sip account and see what happens ? It’s most likely a config issue.

No you dont. You can use Ztdummy

Yes !!

Wrong. You do username:pass@provider.com

If you want the calls coming in to go to a specific extension then you would add to the end of the register statement a forward slash and the extension that you want the call to default to. i.e. username:pass@provider.com/1234

Can you make outbound calls ? Also in the CLI try typing SIP SHOW PEERS and see what comes up. Again seems to be a config issue. Also make sure you are using the Answer command.

[quote=“mad071”]
Help anyone , please?

Greetings

mad071[/quote]

Your welcome :smile:

5

Mad071 Here!
Thanks for response :smile:
It is solved and clear now to me, so again Big Thanks, we can close this toppic.

First, sorry for such a “primitive” question, and i found it out too, after some 10 hr. of testing
and one sleeples night, that it really is possible and
actually very,very easy, once you get a bit more familliar with asterisk and initiall hook-ups’.

( I just got a bit in panic cause of deadline i have and it did not go as expected, too many
variables, dooing asterisk and VoIP 1st time , so i screamed for help.)

TO SUMMARIZE MY OWN QESTIONS/ANSWERS (for newbees in * like me )

( sorry if i repeat above statements,those below are made on my own expirience,and it took only 1 day of playing with asterisk …)

  • Asterisk is really GREAT !!!
  • You dont need audio card to playback any messages ( gsm,wav,mp3 or whatever * can…)
    -You dont need any softphone to auto-answer calls ( and then doo whatever you want too…)
    -You dont need any SPECIAL hardware ( good linux machine with netwoork card will do VoIP as much as your bandwidth /VoIP provider allow you. )
  • depending on SIP provider, YES,you canregistrate more than 1 SIP client/nr to same ip/hostname ( again,no softphone/phone at all connected to * )
  • Same Voiceboxes/menus/audiomessages are simoultaneouslly available on all your SIP
    registraded nr’s ( if you make such config)
    -busytone problems - check any NAT/firewalling in your net and ask your provider too!
    if they do any funny stuff between you and www! ( really, it is worth of try )
    AND, if you dont want DIAL-OUT, costs are more than <minimal/per Year! (10 euro per leased official PHONE NR./line here in europe ( Holland) ).

Well if you only need a call+answer center with extras like “Leave your message after BEEP/press this and that to do THIS-AND-THAT” , this is IT!
It can’t get cheaper, seriouslly. Leasing any lines from post for pure call-in+prerecorded-helpdesk-response-and-leave-your-request-after-beep-tone-we-call-you-back,
and buying expensive hardware to do it is obsolete with asterisk and good internet connection.

What i did/use w. asterisk now ( 2nd day ) :

  • LinuxFromScratch for 64Bit/multilib ( 2 Dual Core Opterons, 2 GB mem , RAID, – well it does not get better than this :wink: )
  • latest asterisk asterisk-1.2.11 + addons + sounds
    ( important maybe - use latest SOX version - sox: v12.18.2 for creating your own gsm/wav messages )
  • And lots of software - webserver,sql,vsftpd,aduio/video streaming server, and now asterisk on same machine to integrate those streams into audiomessages for phone callers.
    Like "press 105 to listen to XXXYYYZZZ …
  • WAV 16 Bit , 8KHz, signed linear is faaaar beter than free gsm for linux. ( gsm works perfect and has less bandwithd ( MUCH LESS ~ 1/10) than wav ,but it gets crispy and
    psychoacoustics are not soo god.)

And it works like clock!
Also , compilling asterisk on 64Bit for 32 bit multilib LinuxOS (codecs…codecs… please port
them all …) went flawlessly :smile:

Ok. I stop now, and thanks for all the help.

1 more happy asterisk user is here :smile:
( does not mean i wont come back with more questions :wink: )

Greetings

mad071