PBXs
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FreePBX / PBX in a Flash (SIP)
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 (one of our multiple servers, you can choose the one closer to your location) secret=***** (password associated with the Main or Sub-account) type=peer username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) disallow=all allow=ulaw ; allow=g729 ; uncomment if you purchased g.729 from Digium fromuser=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) trustrpid=yes sendrpid=yes insecure=invite qualify=yes
Register String: youraccountnumber:[email protected]:5060 (i.e. 123456:[email protected]:5060)
Please note: Some customers have needed to change the fromuser 6 digit account number to their DID number for outgoing CallerID to be displayed as desired.
FreePBX / PBX in a Flash (IAX2)
Fill the blanks with your information, please note that the images above are just examples.
type=friend username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) secret=***** (password associated with the Main or Sub-account) context=from-trunk host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location) disallow=all allow=ulaw insecure=port,invite requirecalltoken=no qualify=yes
Register String: youraccountnumber:[email protected]: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
[general] register => 100000:[email protected]:5060 [voipms] canreinvite=no context=mycontext host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location) secret=johnspassword ;your password type=peer username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) disallow=all allow=ulaw ; allow=g729 ; Uncomment if you support G729 fromuser=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) trustrpid=yes sendrpid=yes insecure=invite nat=yes
- 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
[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 (one of our multiple servers, you can choose the one closer to your location) type=peer disallow=all allow=ulaw ; allow=g729 ; uncomment if you support g729 nat=yes
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:[email protected] [voipms] type=friend username=100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub) secret=johnspassword ;your password context=mycontext host=atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location) 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 #. This example uses newyork.voip.ms as a primary route and chicago.voip.ms as a backup route. You can use whichever hosts you prefer, so long as you keep them consistent in the configuration.
configure terminal voice service voip ip address trusted list ipv4 74.63.41.218 !Current IP address for newyork.voip.ms at the time of this writing. ipv4 173.208.83.50 !Current IP address for chicago.voip.ms at the time of this writing ip address trusted call-block cause not-in-cug 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 realm voip.ms registrar 1 dns:newyork.voip.ms auth-realm voip.ms !Pick your preferred first server registrar 2 dns:chicago.voip.ms auth-realm 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... no voice-class sip block 180 no voice-class sip block 181 no voice-class sip block 183 dtmf-relay rtp-nte no vad voice-class sip bind media source-interface GigabitEthernet0/0 !Use your internet-facing interface here voice-class sip bind control source-interface GigabitEthernet0/0 !Use your internet-facing interface here !This dial peer is for outgoing calls and matches anything. !Finish dialing with a # to immediately route the call. !This is the first-priority dial peer for if your first-priority registrar server is available dial-peer voice 2 voip destination-pattern T no voice-class sip block 180 no voice-class sip block 181 no voice-class sip block 183 dtmf-relay rtp-nte session protocol sipv2 session transport udp session target dns:newyork.voip.ms no vad codec g711ulaw voice-class sip bind media source-interface GigabitEthernet0/0 !Use your internet-facing interface here voice-class sip bind control source-interface GigabitEthernet0/0 !Use your internet-facing interface here preference 1 !This dial peer is the same as #2, but lower preference. !This dial peer will be used for outgoing calls in case the first server is unavailable. !If ANY response (failure or success) is received from the first server, this dial peer will not be used. dial-peer voice 3 voip destination-pattern T no voice-class sip block 180 no voice-class sip block 181 no voice-class sip block 183 dtmf-relay rtp-nte session protocol sipv2 session transport udp session target dns:chicago.voip.ms no vad codec g711ulaw voice-class sip bind media source-interface GigabitEthernet0/0 !Use your internet-facing interface here voice-class sip bind control source-interface GigabitEthernet0/0 !Use your internet-facing interface here preference 2 end copy run start
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.
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 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 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:
Known Issues
RTP Time Out when a person leaves a Voicemail resulting in only a 60 second voicemail being possible.
Fix: Please add the following line to your record_waste_resources.xml or vars.xml file this will make RTP keep alive packets to continue to be sent while recordings are made.
<action application="set" data="record_waste_resources=true"/>
PBXes.org
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.
Nortel/Avaya BCM 450 and BCM50 R6
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. BCM Configuration
