Toronto Asterisk Users' Group

Combustion · 0 users

Cable Internet -> (Public IP @A1) WAN [Dlink DI724U] LAN(Private IP @B1) ->

WAN (Private IP @B2)  [ Linksys WRTSL54GS] LAN(Private IP @C1) 
                   -> Home PC (Private IP @C2)  running GoogleTalk Client 1.0.0.104
                   -> ATA     (Private IP @C3)  with one analogue phone

root@OpenWrt:/# /opt/bin/ipkg install \
  http://ipkg.nslu2-linux.org/feeds/optware/ddwrt/cross/stable/asterisk14_...
root@OpenWrt:/# /opt/bin/ipkg install \
  http://ipkg.nslu2-linux.org/feeds/optware/ddwrt/cross/stable/asterisk14-...
root@OpenWrt:/# /opt/bin/ipkg install \
  http://ipkg.nslu2-linux.org/feeds/optware/ddwrt/cross/stable/asterisk14-...
root@OpenWrt:/# /opt/bin/ipkg install \
  http://ipkg.nslu2-linux.org/feeds/optware/ddwrt/cross/stable/asterisk14-...
root@OpenWRT:/# /opt/bin/ipkg install \
  http://ipkg.nslu2-linux.org/feeds/optware/ddwrt/cross/stable/iksemel_1.2... 

root@OpenWrt:/opt/etc/asterisk# asterisk -r
Asterisk 1.4.8, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail          s.
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 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.8 currently running on OpenWrt (pid = 28709)
Verbosity is at least 11
    -- Remote UNIX connection
OpenWrt*CLI> ?
!           abort       add         agi         cdr         clear       convert
core        database    debug       dialplan    dnsmgr      dont        dump
extensions  feature     file        group       gtalk       help        http
iax2        include     indication  init        jabber      keys        load
local       logger      manager     module      moh         no          originate
reload      remove      restart     rtcp        rtp         save        say
set         show        sip         soft        stop        stun        udptl
unload      voicemail
OpenWrt*CLI>

[general] 
debug=no                      ;;Turn on debugging by default.    
autoprune=no                  ;;Auto remove users from buddy list.
autoregister=no               ;;Auto register users from buddy list.

[asterisk]                    ;;label
type=client                   ;;Client or Component connection
serverhost=talk.google.com    ;;Route to server for example,  talk.google.com
username=username@gmail.com   ;;Username with optional roster.
secret=YOUSECRET              ;;Password
port=5222                     ;;Port to use defaults to 5222
usetls=yes      ;;TLS is required by talk.google.com, you'll get a 'socket read error' without 
usesasl=yes     ;;Use sasl or not
buddy=buddyusername@gmail.com  ;;Manual addition of buddy to list.
statusmessage="This is an Asterisk server" 
timeout=100 

[general]
context=google-in        ;;Context to dump call into
allowguest=yes           ;;Allow calls from people not in
                         ;;list of buddy
bindaddr=Private IP @B2  ;;These two settings are very critical for getting 
externip= Public IP @A1  ;; gtalk audio with Asterisk server behind NAT

[guest]                  ;;special account for options on guest account
disallow=all
allow=ulaw
context=google-in

[buddy]
username=buddyusername@gmail.com  ;;username of the peer your
                                  ;;calling or accepting calls from
disallow=all
allow=ulaw
context=google-in
connection=asterisk    ;;client or component in jabber.conf
                       ;;for the call to leave on.

...
[google-in]
exten => s,1,NoOp( Call from Gtalk )
exten => s,n,Set(CALLERID(name)="From Google Talk")
exten => s,n,Dial(SIP/my_sip_phones)
...
[default]
...
exten => 100,1,JabberStatus(asterisk,buddyusername@gmail.com,STATUS)
exten => 100,2,GotoIf($[${STATUS}<6]?3:5)
exten => 100,3,JABBERSend(asterisk,buddyusername@gmail.com, Hi Buddy  ${CALLERID(name)} at extension ${CALLERID(num)} is calling you!)
exten => 100,4,Dial(gtalk/asterisk/buddyusername@gmail.com)
exten => 100,5,voicemail(100,u)

; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=1650   ;; Googletalk is using low RTP Port numbers than usual Asterisk server
rtpend=4560     ;; Please get the proper number by set 'Jabber debug' on asterisk  
                ;;console, and start gtalk to asterisk, then looking for the jabber 
                ;;negotiations of  RTP ports 
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)

Toronto Asterisk Users' Group

Configuring GoogleTalk with Asterisk 1.4.8 on OpenWRT whiterussian 0.9