VoIP.ms Wiki - User contributions [en]

Talk:PBXs

2012-11-02T18:30:38Z

Upcraft: BCM

== 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!

== Nortel BCM ==
I have added some basic setup instructions for a Nortel/Avaya BCM 450/50 system.

NortelBCM

2012-11-02T18:26:10Z

Upcraft: Created page with "To configure the Nortel BCM you must have VOIP Trunk or SIP Trunk keycodes. The VOIP Trunk (H323) keycode is a bit more expensive and allows both H323 and SIP type trunks. The S..."

To configure the Nortel BCM you must have VOIP Trunk or SIP Trunk keycodes.

The VOIP Trunk (H323) keycode is a bit more expensive and allows both H323 and SIP type trunks.
The SIP Trunk keycode only allows for SIP trunks.

Your number of trunk licenses will determine the maximum number of simultaneous IP trunks that can be used.

H323 Trunks are commonly used for linking together different BCM systems to form a private network. Usually a BCM in a branch office or another building is linked using H323 trunks. Private networking can also be done with SIP trunks but the support for H323 has traditionally been better integrated and has a longer history.

To check the keycodes on your system Log in with Business Element Manager and view Configuration->System->Keycodes
H323/SIP keycodes are labeled: VoIP Gateway Trunk
SIP keycodes are labeled: SIP Trunk
[[File:BCMKeycodes.jpg]]

To configure the Trunk
go to Configuration->Resources->IP Trunks->General
Under IP Trunk Settings I have it set for:
Forward redirected OLI: First Redirect
Remote capability MWI: checked
Send name display: checked
Ignore in-band DTMF in RTP: unchecked
[[File:BCMGeneralIPTrunkSettings.jpg]]

To configure your SIP trunk start by creating the Account
Go to Configuration->Resources->IP Trunks->SIP Trunking
Click on Add. Choose No Template...
Name: A short name you will see in a few places to identify the account (No spaces, only certain special characters are allowed)
Description: A longer description you can use to be more specific.
SIP Domain: use the server name of your voip.ms account. eg: chicago.voip.ms or seattle.voip.ms
Registration required: checked
SIP username: this is your VOIP.ms account number or subaccount
Password: Your account or subaccount password. PRESS TAB when you type in the password, do not just click on OK. You must press tab or click on another box to make the password confirmation screen come up.

[[File:BCMAddSIPAccount.jpg]]

Once added you can choose the account and below are more options to configure.
On the Basic screen the Proxy can be set to the server name of your server.
The registrar is also the server name for your server
Set the ports to 5060
You likely do not need an outbound proxy. (Unless you have a fancy firewall at your office with a SIP proxy feature)

[[File:BCMAccountBasic.jpg]]

Under Advanced I have:
Enable local NAT compensation: Unchecked (You can try turning this on if you have STUN setup in Configuration->System->IP Subsystem if your router doesn't forward SIP sessions properly)
Enable media relay: checked
Use maddr in R-URI: unchecked
Use maddr in Contact: unchecked
Support 100rel: checked
Allow UPDATE: checked
Use Null IP to hold: checked
Use user=phone: unchecked
Force E164 international settings: unchecked
Enable SDP OPTIONS query: unchecked
Allow REFER: checked
Support Replaces: checked
Enable Connected Identity: checked
Standard SIP Caps Exchange: unchecked

NAT Pinhole Maintenance, Signaling Method: NONE, Signaling interval: 30
Session timer, Session refresh method: Disable
Active call limit: 0 (You can set this to 2 if you have an unlimited residential account to limit you to 2 channels)

ITSP association method: From header domain match
Outbound Called characters to absorb: 0
Inbound Called prefix to prepend: blank
Authentication realm: blank

[[File:BCMAccountAdvanced.jpg]]

Under User Accounts
There should be a parent user account here already from adding the account
You can add more accounts. I have not played around with this but you should be able to add multiple subaccounts in this section if you have multiple DID accounts you want to setup in your system.
The description of the parent account will be blank, you can leave it blank or add something if you like.
You can override the caller ID info here. You may need to experiment with these to get your preferred results on what numbers show up on inbound and outbound calls.

The most useful part of the User Accounts window will be the registration status. This is where you look to see if your BCM has connected to the voip.ms service. You can also check the corresponding registration status on your voip.ms online account page.

[[File:BCMAccountsUserAccounts.jpg]] The username and password is just an example so it gives me a forbidden message when trying to register.
[[File:BCMAccountsUserAccountsRegd.jpg]] This is a real username that is working.

Once registered you should be able to call into your system. Without any routing info it should ring on your prime set. You can use the BCM Monitor utility to view the inbound call in the Line Monitor screen and once you pick up you can view the details of the RTP connection in the RTP Sessions tab.

For more info on further setup tasks like setting up call routing, destination digits, and other BCM specific configuration info you can check the online documentation at support.avaya.com
There are some example configurations for other IP service providers that are similar to using voip.ms

For example check this one:
[https://downloads.avaya.com/css/P8/documents/100155746 Avaya application tech note for implementing SIP trunking with BT]

File:BCMAccountsUserAccountsRegd.jpg

2012-11-02T18:20:45Z

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File:BCMAccountsUserAccounts.jpg

2012-11-02T18:18:04Z

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File:BCMAccountAdvanced.jpg

2012-11-02T18:08:25Z

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File:BCMAccountBasic.jpg

2012-11-02T18:01:38Z

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File:BCMAddSIPAccount.jpg

2012-11-02T17:57:12Z

Upcraft: uploaded a new version of "File:BCMAddSIPAccount.jpg"

File:BCMAddSIPAccount.jpg

2012-11-02T17:55:00Z

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File:BCMGeneralIPTrunkSettings.jpg

2012-11-02T17:43:50Z

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File:BCMKeycodes.jpg

2012-11-02T17:28:33Z

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PBXs

2012-11-02T17:15:12Z

Upcraft:

==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]]
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==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]]
<div style="width:100%;overflow:hidden;clear:both"></div>
 

==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]]

==Nortel/Avaya BCM 450 and BCM50 R6==

[[File:Avaya.jpg|300px|thumb|left|Avaya BCM]]
 
Avaya BCM450 (formerly Nortel BCM 450) and the BCM50 with VOIP or SIP Trunk keycode license.

The BCM system is a popular legacy Nortel phone system that uses classic Nortel "Meridian" M and T series digital sets and their Unistim IP phones.
[[NortelBCM|BCM Configuration]]
<div style="width:100%;overflow:hidden;clear:both"></div>
 
[[category:PBXes]]

File:Avaya.jpg

2012-11-02T17:11:15Z

Upcraft: