Asterisk 1.2.14 svn rev 48468, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.14 svn rev 48468 currently running on asterisk1 (pid = 27416)
Verbosity is at least 3
in: additional_myapplication_sip.conf
[9120182351]
type=friend
username=9120182351
accountcode=9120182351
regexten=9120182351
callerid=12345678
amaflags=billing
secret=******
nat=yes
dtmfmode=RFC2833
qualify=yes
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
host=dynamic
context=myapplication-1
mailbox=12345678
regseconds=0
cancallforward=yes
in sip.conf
;++++++++++++++++++++++++++++++++++++++++++++++++++++++
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.
[general]
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
disallow=all
allow=g729
allow=g723
allow=ulaw
allow=alaw
allow=gsm
;allow=ilbc
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68
nat=route
externip=*.*.*.*
#localnet=192.168.0.0/255.255.255.0
qualify=yes
notifymimetype=application/simple-message-summary
; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
#include additional_myapplication_sip.conf
in voicemail.conf
;+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++==
[general]
#include vm_general.inc
#include vm_email.inc
[default]
12345678 => 2386,test301,myemail@yahoo.com,attach=no|sayduration=yes|forcename=yes|forcename=yes|forcegreetings=no|review=yes|operator=yes