Asterisk SIP Trunk to Broadsoft behind an Edgemarc
4550 using Transparent Proxy
NOTE** I use the SBC with IP of 10.x.x.60 for labvoip.cableone.net platform as outbound
proxy. I do not set this globally as other calls try to go back out the
proxy/SBC and never reach the internal extensions. So I use this in the trunk
register => line for inbound calls and the [6026356917] for outbound calls.
Edgemarc 4550 setup:
·
LAN network 192.168.1.0/24
·
Edgemarc LAN IP 192.168.1.1
·
WAN network 10.137.128.0/24
·
Edgemarc WAN IP 10.x.x.18
·
Manually entered my DNS
·
DHCP on and using defaults
·
I have allowed http and ssh in firewall settings
·
I have port forward 2222 from WAN to port 22 for
PBX IP (192.168.1.10) in LAN. Just for management outside the LAN.
·
I have port forward 8080 from WAN to port 80 for
Phone IP (192.168.1.20) in LAN. Just for management outside the LAN.
·
Transparent Proxy is on under NAT
Asterisk server network setup
·
LAN IP 192.168.1.10/24
·
Gateway and DNS set to 192.168.1.1
Asterisk 11.15.0 sip.conf
;sip.conf
[general]
nat=yes
;used to register with broadsoft for inbound calls
context=public ; Default context for
incoming calls. Defaults to 'default'
allowoverlap=no ; Disable overlap dialing
support. (Default is yes)
udpbindaddr=192.168.1.10 ; IP address to bind UDP listen socket
to (0.0.0.0 binds to all)
tcpenable=no ; Enable server for
incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to
bind to (0.0.0.0 binds to all interfaces)
transport=udp ; Set the default
transports. The order determines the
primary
default transport.
srvlookup=yes ; Enable DNS SRV lookups on
outbound calls
secret=password
context=public
host=dynamic
allow=all
dtmfmode=rfc2833
;this
is for polycom sip phone
[6026356916]
type=friend
callerid=6026356916
secret=password
context=public
host=dynamic
allow=all
dtmfmode=rfc2833
; this is for asterisk to brs sip trunk using pilot number for out
bound calls
[6026356917]
type=peer
user=phone
username=6026356917
host=labvoip.xxx.xxx
fromdomain=labvoip.xxx.xxx
realm=labvoip.xxx.xxx
secret=password
fromuser=602635xxxx
outboundproxy=10.x.x.60:5060
;Disable canreinvite if you are behind a NAT
canreinvite=no
context=public
Asterisk 11.15.0 extensions.conf
;
extensions.conf - the Asterisk dial plan
;
[public]
;used
to pass numbers dialed to SIP Trunk
exten
=> _X.,1,Dial(SIP/${EXTEN}@6026356917)
exten
=> _X.,n,Hangup
;used
when trunk pilot DID of 6026356917 is called it rings ext 6026356916
exten
=> 6026356917,1,Answer
exten
=> 6026356917,n,Dial(SIP/6026356916,20)
exten
=> 6026356917,n,Hangup
;used
when DID 6026356916 is called it rings ext 6026356916
exten
=> 6026356916,1,Answer
exten
=> 6026356916,n,Dial(SIP/6026356916,20)
exten
=> 6026356916,n,Hangup
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console
interface for demo
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G2 ; Trunk
interface
;
;
Note the 'G2' in the TRUNK variable above. It specifies which group (defined
; in
chan_dahdi.conf) to dial, i.e. group 2, and how to choose a channel to use
; in
the specified group. The four possible options are:
;
; g:
select the lowest-numbered non-busy DAHDI channel
; (aka. ascending sequential hunt group).
; G:
select the highest-numbered non-busy DAHDI channel
; (aka. descending sequential hunt group).
; r:
use a round-robin search, starting at the next highest channel than last
; time (aka. ascending rotary hunt group).
; R:
use a round-robin search, starting at the next lowest channel than last
; time (aka. descending rotary hunt group).
I could really help getting asterisk to work with cisco phones.
ReplyDeleteI could really help getting asterisk to work with cisco phones.
ReplyDeleteDo you need help with sip phones and asterisk? What phones are you using? What version of asterisk?
Delete