VoIP.ms Wiki - User contributions [en]

PBXs

2011-06-30T03:01:23Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start

PBXs

2011-06-30T03:01:09Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start

PBXs

2011-06-30T03:00:56Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start

PBXs

2011-06-30T02:22:30Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start

PBXs

2011-06-30T02:22:14Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start

PBXs

2011-06-30T02:21:59Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start

PBXs

2011-06-30T02:17:23Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start

PBXs

2011-06-30T02:17:03Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===

<nowiki>
[mycontext]
include => voipms-outbound
include => voipms-inbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start

PBXs

2011-06-30T02:16:10Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===

<nowiki>
[mycontext]
include => voipms-outbound
include => voipms-inbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===

<nowiki>
[mycontext]
include => inbound
include => outbound

[outbound]
exten => _1NXXNXXXXXX,1,Dial(IAX2/109799@voipms/${EXTEN})
exten => _1NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(IAX2/109799@voipms/${EXTEN})
exten => _011.,n,Hangup()
exten => _00.,1,Dial(IAX2/109799@voipms/${EXTEN})
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start

PBXs

2011-06-30T02:15:47Z

Dereksky: /* extensions.conf */

==FreePBX / Trixbox / PBX in a Flash (SIP)==

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
canreinvite=nonat 
<nowiki>; nat=yes ; uncomment if behind nat</nowiki> 
context=from-trunk 
host=atlanta.voip.ms 
secret=johnspassword ;your password 
type=friend 
username=100000 ;your account 
disallow=all 
allow=ulaw 
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki> 
fromuser=100000 ;your account 
trustrpid=yes 
sendrpid=yes 
insecure=port,invite 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:5060
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==FreePBX / Trixbox / PBX in a Flash (IAX2)==

https://www.voip.ms/m/samples/images/freepbxiax.gif
[[File:Pbx-left.jpg|left]]

{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
type=friend 
username=100000 ;your account 
secret=johnspassword ;your password 
context=from-trunk 
host=atlanta.voip.ms 
disallow=all 
allow=ulaw 
insecure=port,invite 
requirecalltoken=no 
qualify=yes 
|}

Register String
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
|
100000:johnspassword@atlanta.voip.ms:4569
|}
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==Asterisk (SIP)==

===sip.conf===

<nowiki>
[general]
register => 100000:johnspassword@atlanta.voip.ms:5060

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=friend
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=port,invite
; nat=yes ; Uncomment this if your box is behind a NAT
</nowiki>

===extensions.conf===
<nowiki>
[mycontext]
; Make sure to include inbound prior to outbound cause the _NXXNXXXXXX handler will fetch the incoming call and create a loop
include => voipms-outbound
include => voipms-inbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk IP Auth. (SIP)==

===sip.conf===

Note: You'll need to create a sub account to use IP Auth

<nowiki>
[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=friend
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
insecure=port,invite
; nat=yes ; uncommment if behind a nat
</nowiki>

===extensions.conf===

<nowiki>
[mycontext]
include => voipms-outbound
include => voipms-inbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>
 

==Asterisk (IAX2)==

===iax.conf===

<nowiki>
register => 100000:johnspassword@atlanta.voip.ms

[voipms]
type=friend
username=100000 ;your account
secret=johnspassword ;your password
context=mycontext
host=atlanta.voip.ms
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support it
insecure=port,invite
requirecalltoken=no
</nowiki>

===extensions.conf===

<nowiki>
[mycontext]
include => inbound
include => outbound

[outbound]
exten => _1NXXNXXXXXX,1,Dial(IAX2/109799@voipms/${EXTEN})
exten => _1NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(IAX2/109799@voipms/${EXTEN})
exten => _011.,n,Hangup()
exten => _00.,1,Dial(IAX2/109799@voipms/${EXTEN})
exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]
exten => 7863643011,1,Answer() ;your DID
</nowiki>

==Cisco IOS==

===SIP Trunk (Username/Password Authentication)===
For the configuration below to work, you must have DNS name lookups properly configured on your router.
The example below is based on IOS 15.1(3)T. Minor adjustments may be necessary for ealier IOS revisions.
Note that the dial plan shown below will allow you to dial anything and will pass those digits directly to voip.ms, but will take a few seconds to route the call unless you finish the number with a #.

configure terminal

voice service voip
gcid
clid substitute name
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
transport switch udp tcp
asserted-id ppi
localhost dns:dns.name.of.your.device
midcall-signaling passthru
no call service stop

sip-ua
credentials username your_account password 0 your_password realm voip.ms
authentication username your_account password 0 your_password
registrar 1 dns:newyork.voip.ms !Pick your preferred first server
registrar 2 dns:montreal.voip.ms !Pick the next best here
!You can configure up to 6 registrar servers for fault-tolerance

!This dial peer will match all incoming calls for DIDs in the range (555)555-1000 to (555)555-1999
dial-peer voice 1 voip
incoming called-number 5555551...
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte

!This dial peer is for outgoing calls and matches anything.
!Finish dialing with a # to immediately route the call.
dial-peer voice 2 voip
destination-pattern T
voice-class sip asserted-id ppi
no voice-class sip block 180
no voice-class sip block 181
no voice-class sip block 183
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
session protocol sipv2
session transport udp
session target sip-server
end
copy run start