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PBXs

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(FreePBX / PBX in a Flash (IAX2))
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==FreePBX / PBX in a Flash (SIP)==
==FreePBX / PBX in a Flash (SIP)==
-
https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif
+
https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif  
-
[[File:Pbx-left.jpg|left]]
+
-
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
+
[[File:Siptrunk.png]]
-
|
+
-
canreinvite=nonat<br>
+
-
<nowiki>; nat=yes ; uncomment if behind nat</nowiki><br>
+
-
context=from-trunk<br>
+
-
host=atlanta.voip.ms<br>
+
-
secret=johnspassword ;your password<br>
+
-
type=friend<br>
+
-
username=100000 ;your account<br>
+
-
disallow=all<br>
+
-
allow=ulaw<br>
+
-
<nowiki>; allow=g729 ; uncomment if you purchased g.729 from Digium</nowiki><br>
+
-
fromuser=100000 ;your account<br>
+
-
trustrpid=yes<br>
+
-
sendrpid=yes<br>
+
-
insecure=port,invite<br>
+
-
qualify=yes<br>
+
-
|}
+
 +
'''''Fill the blanks with your information, please note that the images above are just examples.'''''
-
Register String
+
canreinvite=nonat
-
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
+
;nat=yes ;uncomment if behind a nat
-
|
+
context=from-trunk
-
100000:johnspassword@atlanta.voip.ms:5060
+
host=atlanta.voip.ms
-
|}
+
secret=
-
<div style="width:100%;overflow:hidden;clear:both"></div>
+
type=peer
-
<br><br><br><br><br>
+
username=
 +
disallow=all
 +
allow=ulaw
 +
; allow=g729 ; uncomment if you purchased g.729 from Digium
 +
fromuser=
 +
trustrpid=yes
 +
sendrpid=yes
 +
insecure=invite
 +
qualify=yes
 +
 
 +
Register String:
 +
youraccountnumber:yourpassword@atlanta.voip.ms:5060
==FreePBX / PBX in a Flash (IAX2)==
==FreePBX / PBX in a Flash (IAX2)==
https://www.voip.ms/m/samples/images/freepbxiax.gif
https://www.voip.ms/m/samples/images/freepbxiax.gif
-
[[File:Pbx-left.jpg|left]]
 
-
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
+
[[File:Iaxtrunk.png]]
-
|
+
-
type=friend<br>
+
-
username=100000 ;your account<br>
+
-
secret=johnspassword ;your password<br>
+
-
context=from-trunk<br>
+
-
host=atlanta.voip.ms<br>
+
-
disallow=all<br>
+
-
allow=ulaw<br>
+
-
insecure=port,invite<br>
+
-
requirecalltoken=no<br>
+
-
qualify=yes<br>
+
-
|}
+
 +
'''''Fill the blanks with your information, please note that the images above are just examples.'''''
-
Register String
+
type=friend
-
{|style="border-collapse: collapse; border:1px solid #000; width:500px"
+
username=
-
|
+
secret=
-
100000:johnspassword@atlanta.voip.ms:4569
+
context=from-trunk
-
|}
+
host=atlanta.voip.ms
-
<div style="width:100%;overflow:hidden;clear:both"></div>
+
disallow=all
-
<br><br><br><br><br>
+
allow=ulaw
 +
insecure=port,invite
 +
requirecalltoken=no
 +
qualify=yes
 +
 
 +
Register String:
 +
youraccountnumber:yourpassword@atlanta.voip.ms:4769
==Asterisk (SIP)==
==Asterisk (SIP)==
Line 73: Line 61:
host=atlanta.voip.ms
host=atlanta.voip.ms
secret=johnspassword ;your password
secret=johnspassword ;your password
-
type=friend
+
type=peer
username=100000 ;your account
username=100000 ;your account
disallow=all
disallow=all
Line 81: Line 69:
trustrpid=yes
trustrpid=yes
sendrpid=yes
sendrpid=yes
-
insecure=port,invite
+
insecure=invite
-
; nat=yes ; Uncomment this if your box is behind a NAT
+
nat=no
</nowiki>
</nowiki>
Line 122: Line 110:
context=mycontext
context=mycontext
host=atlanta.voip.ms
host=atlanta.voip.ms
-
type=friend
+
type=peer
disallow=all
disallow=all
allow=ulaw
allow=ulaw
; allow=g729 ; uncomment if you support g729
; allow=g729 ; uncomment if you support g729
-
insecure=port,invite
+
nat=no
-
; nat=yes ; uncommment if behind a nat
+
</nowiki>
</nowiki>
Line 277: Line 264:
<br><br>
<br><br>
trixbox Community Edition began in 2004 as the massively popular Open Source IP-PBX project named Asterisk@Home. Since then, it has grown into the world's most popular distribution of Asterisk with over 65,000 downloads per month. trixbox is known for its flexibility to satisfy the needs of custom deployments and will continue to be FREE (as in beer and freedom).  
trixbox Community Edition began in 2004 as the massively popular Open Source IP-PBX project named Asterisk@Home. Since then, it has grown into the world's most popular distribution of Asterisk with over 65,000 downloads per month. trixbox is known for its flexibility to satisfy the needs of custom deployments and will continue to be FREE (as in beer and freedom).  
-
 
+
'''Trixbox has not been maintained since June 2010. Customers should look for alternatives.'''
[[Trixbox|Trixbox Configuration]]
[[Trixbox|Trixbox Configuration]]
<div style="width:100%;overflow:hidden;clear:both"></div>
<div style="width:100%;overflow:hidden;clear:both"></div>

Revision as of 18:45, 4 August 2011

Contents

FreePBX / PBX in a Flash (SIP)

freepbxsiptrunk.gif

Siptrunk.png

Fill the blanks with your information, please note that the images above are just examples.
canreinvite=nonat
;nat=yes ;uncomment if behind a nat
context=from-trunk
host=atlanta.voip.ms
secret=
type=peer
username=
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you purchased g.729 from Digium
fromuser=
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes
Register String:
youraccountnumber:yourpassword@atlanta.voip.ms:5060

FreePBX / PBX in a Flash (IAX2)

freepbxiax.gif

Iaxtrunk.png

Fill the blanks with your information, please note that the images above are just examples.
type=friend
username=
secret=
context=from-trunk
host=atlanta.voip.ms
disallow=all
allow=ulaw
insecure=port,invite
requirecalltoken=no
qualify=yes
Register String:
youraccountnumber:yourpassword@atlanta.voip.ms:4769

Asterisk (SIP)

sip.conf

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

[voipms]
canreinvite=no
context=mycontext
host=atlanta.voip.ms
secret=johnspassword ;your password
type=peer
username=100000 ;your account
disallow=all
allow=ulaw
; allow=g729 ; Uncomment if you support G729
fromuser=100000 ;your account
trustrpid=yes
sendrpid=yes
insecure=invite
nat=no

extensions.conf

[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






Asterisk IP Auth. (SIP)

sip.conf

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

[voipms]
canreinvite=nonat
context=mycontext
host=atlanta.voip.ms
type=peer
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
nat=no


extensions.conf

[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






Asterisk (IAX2)

iax.conf

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


extensions.conf

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






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






Talkswitch

Talkswitch



Unlike other small business telephone systems, TalkSwitch is easy to install, saving you time and money; you can even do it yourself. And with its easily configured settings, moving employees or changing the way your phone system handles calls is a snap.

Talkswitch Configuration






Trixbox

Trixbox



trixbox Community Edition began in 2004 as the massively popular Open Source IP-PBX project named Asterisk@Home. Since then, it has grown into the world's most popular distribution of Asterisk with over 65,000 downloads per month. trixbox is known for its flexibility to satisfy the needs of custom deployments and will continue to be FREE (as in beer and freedom). Trixbox has not been maintained since June 2010. Customers should look for alternatives. Trixbox Configuration






3CX Phone System

3CX Phone System



3CX Phone System for Windows is a software-based IP PBX that replaces a proprietary hardware PBX / PABX. 3CX’s IP PBX has been developed specifically for Microsoft Windows and is based on the SIP standard – making it easier to manage and allowing you to use any SIP phone (software or hardware).

3CX Phone System Configuration






SIPfoundry

Sipfoundry-logo.png

SIPfoundry is the home of the sipXecs open source unified communications and collaboration project. If you are looking to replace your aging (IP) PBX with an all software modern communications solution that scales to mid-size and large enterprise, look no further. We are community dedicated to the development of SIP and XMPP based communications solutions. Founded in 2004 as an independent 501(3)c not-for-profit corporation based in Massachusetts, SIPfoundry is governed by an independent board of directors. SIPfoundry is open an invites all interested parties to cooperate and collaborate. While the sipXecs project is the largest active project at SIPfoundry, we are open to make available our infrastructure to other interested projects. A commercial enterprise edition of the sipXecs solution is available from eZuce with full support, software maintenance, training, indemnification, and other services.

To learn how to configure sipXecs to work with voip.ms, follow this 10 minute guide here:

http://blog.myitdepartment.net/?p=191






PBXes.org

PBXes



PBXes is a hosted PBX running like a voicemail service or an email server in a data center. Like a classical PBX it serves as the bridge between your organisation (the extensions) and the public telephone network (the trunks). Thus it optimises both your internal and external communications. As trunks any SIP providers worldwide may be used.

PBXes Configuration






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