Newb in diapers with questions

Asterisk Asterisk General
1

I did some searching to no avail before posting so I apologize if this is covered 30 different times in here somewhere.

I have asterisk up and running on an opensuse 10.3 box.

My question is what do I need to do to be able to make and recieve calls.

Not asterisk setup but the other stuff.

I must need some kind of internet DID to receive calls, or so I believe.
I have heard that I do not need anythinng to make calls though. I believe this part to be wrong.

What I am trying to accomplish is to have a sip phone that can accept and send calls.

For starters I would like to have 3 people on hold and talking to a fourth person.
I want to eventually be able to have 9 people on hold and talking to a tenth.

I just do not know the steps to take on this part of the setup. As far as asterisk itself goes, I am doing great on that end.

Can someone give me a kick start in the right direction, maybe a really informative document on this part of the setup?

Thank you in advance for the help,

2

If you want to go all VOIP then what you need is a provider, there are many, many choices and most of them are listed on the wiki.

You need a provider for both incoming and outgoing calls. If you want to have more than one call at a time you will need to specify to the provider how many simultaneous calls you want. Also keep in mind that you need to watch how much bandwith you are consuming for these calls. There are some bandwith calculators on the wiki as well. If you start to stress the internet connection call quality will suffer so its best to over estimate the amount af bandwith required.

http://www.voip-info.org/wiki/view/VOIP+Service+Providers

http://www.voip-info.org/wiki-Bandwidth+consumption

3

ok i got a free number from sipnumber.net for testing purposes and have now ran into problems of a different nature.

I have a phone number of 6414107547.
I am trying to call it and have it ring into my pbx box, play a sound file, and hangup.

Right now it is not calling my pbx box.

This is the information i was given for connecting from the provider.

displayname Mark
username 6414107547
password tomatoes (i changed it to keep it simple for posting)
Domain/Realm sipnumber.net
sip proxy sipnumber.net
out bound proxy outbound.sipnumber.net

This is what I have in my sip.conf file

[general]
context=default ; Default context for incoming calls

[sip_number]
type=peer
context=internal
secret=tomatoes
username=6414107547 ; Authentication user for outbound proxies
fromuser=Mark ; Many SIP providers require this!
fromdomain=sipnumber.net
host=sipnumber.net
usereqphone=yes ; This provider requires “;user=phone” on URI
call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
outboundproxy=outbound.sipnumber.net ; send outbound signaling to this proxy, not directly to the peer
port=80

and this is what I have in my extensions.conf file

[incoming]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,Hangup()

NOTE: All the rest of my conf files are the generic sample ones right now.

i typed module reload into the cli and received this output

NOTICE [9642]: cdr.c:1359 do_reload: CDR simple logging enabled.

can someone help out here? what did i not do or what did i do wrong?
NOTE:
I do not have an actual phone hooked up to the box yet. just trying to connect to it from outside right now.

4

Hi,
you also need a ‘register’ line which I think goes in the general section of sip.conf (I use Freepbx for configuring Asterisk, it’s infinitely easier than manually editing file).

This allows asterisk to log in to the remote system, so that system knows where to route a call to your account.

Typical lines are like:
register => 1040000:ABCDEFGH@sipgate.co.uk/1040000
which is
username:password@provider/username

Some don’t need the /username on the end, some do - without it, the call may try to go to extension 1040000 rather than be seen as a general incoming call. Which works best depends on your dialplan.

Some need username:password:username@…
or it just does not register (seems quite a rare format).
Try the first style, it’s most likely to work.

Are you sure the Port should be 80 in your sip_number block? I’d expect 5060 or something like that, but it is down to the service provider…

5

You are correct. the port should be 5060.

Ill try the rigistering and see if that fixes it.
Ill also check out freepbx.

Thanks for the help

6

ok.

I can authenticate with my sip provider, but I can not call my pbx box.
I know this because when i change 1 letter of my password, when I reload modules i get failure to authenticate message.

i have this in my sip conf file
[general]
context=default
register => 6414107547:tomatoes@proxy.freedigits.net:5060

[SipNumber]
type=user
context=from-pstn
callerid=6414107547
dtmfmode=rfc2283
fromdomain=freedigits.net
host=freedigits.net
insecure=very
secret=tomatoes
user=6414107547
username=6414107547

and i have my extensions file:

[incoming]
exten => s,1,Answer()
exten => s,2,Playback(hello-world)
exten => s,3,Hangup()

This is ALL i have in these files… nothing else.

Can anyone shed some light on what I might be missing.
I have circumvented my router in case that was the problem.

could it be something to do with my linux setup, or would it definately be asterisk.
I just need to know what is wrong or possibly wrong.

Appreciate the help.

7

Is there any firewall between you and the provider? On the linix install itself or your internet connection? If so you need to open the ports on your firewall(s). See the wiki for more info.

8

i did have the firewall running. I have turned that off. I had already bypassed my router so that one isnt a problem.

I can still register successfully but cant call to the box.

9

WOOT!

I have found the problem.

If you will notice in my postings i have this line in the sip.conf file

context=default

in my extensions.conf file i was using incoming.

They did not match so i wasnt getting to it.

I can now receive calls, but this creates my new delima. The sound quality is very poor when I call. The sound is garbled. I have DSL so I shoud not have a problem with just the one call. Could it be that I am using a free number for my testing and that it is their end that is th4e poor quality, or could this be something on my end that I can adjust.

Try my number if you would and ive me feedback on your assumptions based on the sound. The number is 641 410 7547. I have it set up right now to playback the tt-allbusy message.

Thanks for the help folks

10

my guess is its the providers end but it could be yours as well.

What type of comuter are you running asterisk on? I would say minimum should be P4 1.2GHz w\256 MB ram? I would not use any type of virtual machine software (VMware, Zen etc) as the results are frequently very poor.

Try a different provider and remember - you get what you pay for!

11

I had an old computer that was sitting around that im using.

It has a 800 mhz AMD processor with either 256 or 512 mem…cant remember and away from it right now.

This comp is at least 5 years old.

I know well the saying on you get what you pay for. Was holding off paying till everything was set up. Might have to change my thinking on this though.