[Asterisk-Users] Asterisk Sacrifice?

[John Haigh]

Hi Here is my sip configuration with fwd. I would recommend getting a
fwd account (fwd.pulver.com) as it is free.

;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0
; Address to bind to
context = default               ; Default for incoming calls
tos=reliability

register=37526:mypassword at fwd.pulver.com/37526

[fwd]
type=friend
secret=mypassword
username=37526
host=fwd.pulver.com

My extensions.conf

; at the top you will find the globals section...specify a variable
called PHONE1 for sip/fwd..then 
; you can change it out or add another phone variable if you have other
sip phones. 

[globals]

PHONE1=SIP/fwd 

; Extension 1234
; this will dial my soft sip phone x-lite when someone dials 1234...no
ever does though
exten => 1234,1,Playback(transfer,skip)         ; "Please hold while..."
exten => 1234,2,Macro(stdexten,1234,${PHONE1})
exten => 1235,1,Voicemail(u1234)                ; Right to voicemail
exten => 1236,1,Dial(${PHONE1},30)               ; Ring forever
exten => 1236,2,Voicemail(u1234)                ; Unless busy


Also check out John Todd's asterisk conf files. I think they are great
and they helped me get my head around all this. 
http://www.loligo.com/asterisk/current/extensions.conf

John Haigh

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Kelly
McDonald
Sent: Friday, July 04, 2003 8:40 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk Sacrifice?


bk, 

I'm a newbie myself, but I have at least got * working with a sip
provider, although the quality was not to my liking, I was hooking up
with iconnecthere. Here's what I had

in sip.conf:

[iconnecthere]
type=friend
insecure=yes
port=5060
username=xyz
secret=abc
host=natrelay.deltathree.com
dtmfmode=inband
callerid=15408675512
nat=yes

in extensions.conf:

exten => 8500,1,Dial(SIP/15405551212 at iconnecthere)


This was just a test so I could dial 8500 and it would call my home
phone.

Probably have stuff wrong, but it seemed to work.

For the rest, extensions.conf has enough stuff in it that you can go and
make up your own stuff.

HTH,
Kelly

On Fri, 2003-07-04 at 08:23, BK [address only for mailing lists] wrote:
> Hi
> 
> is there any ritual sacrifice a newbie has to perform to be welcome on
> this list?
> 
> I am new to this whole PBX thing in general and Asterisk in 
> particular.
> I had hoped that the community on this list would welcome a newbie
like 
> myself and help me with some answers to my stupid questions, but
somehow 
> it seems to me that nobody likes to respond to somebody who appears to

> be a complete beginner -- too much bother and a risk to have to
explain 
> everything from scratch -- better not answer at all and all that.
> 
> Well, it may appear that way, but I am not a complete idiot. I know a
> lot about mobile switching centres, HLRs, VLRs, IN service nodes, 
> mediation devices, billing and settlement systems etc -- I just don't 
> know much about PSTN and PBXes. I would appreciate it if somebody
could 
> help me out with a few hints on how to set up my Asterisk box, in 
> particular in respect of VoIP as per my last posting.
> 
> thank you very much in advance
> kind regards
> bk
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Kelly McDonald <w4kpm at adelphia.net>

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