Talk:PBXs

2012-02-26T21:09:00Z

Symtry: /* Cisco IOS */

== Cisco IOS ==
I have added a Cisco IOS section for configuration of Cisco voice-enabled routers, using username/password authentication.
The dial plan is extremely basic, but there's plenty of help around the rest of the internet for people to figure out a good dial plan.

The configuration shown should work on any voice-enabled platform running IOS 15.1(3)T and higher, such as the 2800, 3800, 2900, and 3900 series routers.

I just submitted a change for IAX2 configuration, because I wasted a good 4 or 5 hrs screwing around with my FreePBX setup in order to have incoming DIDs work. The problem was that my trunk was not named "voipms"! I know my formatting is a little off, but please ensure that error message I posted and the change is noted for future voip.ms customers. Thanks!

PBXs

2012-02-24T03:08:07Z

Symtry: /* FreePBX / PBX in a Flash (IAX2) */

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

https://www.voip.ms/m/samples/images/freepbxsiptrunk.gif

[[File:Siptrunk.png]]

'''''Fill the blanks with your information, please note that the images above are just examples.'''''

canreinvite=nonat
nat=yes
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)==

https://www.voip.ms/m/samples/images/freepbxiax.gif

[[File: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:4569

'''NOTE''': The trunk name should be set to '''''voipms''''' in lowercase. Otherwise you may have issues with the incoming calls.

If the trunk name is not specifically set to '''''voipms''''', the following error may result on inbound calls: "Call rejected, CallToken Support required."

==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=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=yes
</nowiki>

*Note: As of FreePBX/Asterisk version 2.9.0.7 if ;comments are placed after the config entries as above it may prevent outgoing call's with the end points getting "All circuits are busy". Remove the ;comments and the trunk will send the calls with no errors.

===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=peer
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you support g729
nat=yes
</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(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
</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
 

==Talkswitch==

[[File:TalkSwitch.png|300px|thumb|left|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 PBX|Talkswitch Configuration]]
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==Trixbox==

[[File:Trixbox_logo.jpg|300px|thumb|left|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|Trixbox Configuration]]
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==3CX Phone System==

[[File:3CX Logo.jpg|300px|thumb|left|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|3CX Phone System Configuration]]
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==SIPfoundry==

[[File: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
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==PBXes.org==

[[File:Pbxeshead.png‎|300px|thumb|left|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.org|PBXes Configuration]]
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[[category:PBXes]]

VoIP.ms Wiki - User contributions [en]

Talk:PBXs

PBXs