VoIP.ms Wiki - User contributions [en]

PBXs

2011-07-05T00:27:42Z

Kage1: /* sip.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<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>
|}

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>
<br><br><br><br><br>

==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<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>
|}

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>
<br><br><br><br><br>

==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>

*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>
<br><br><br><br><br>

==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>
<br><br><br><br><br>

==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>
<br><br><br><br><br>

==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
<br><br><br><br><br>

==Talkswitch==

[[File:TalkSwitch.png|300px|thumb|left|Talkswitch]]
<br><br>
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]]
<div style="width:100%;overflow:hidden;clear:both"></div>
<br><br><br><br><br>

==Trixbox==

[[File:Trixbox_logo.jpg|300px|thumb|left|Trixbox]]
<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|Trixbox Configuration]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Devices

2011-06-17T01:41:07Z

Kage1: Added Telco_AC211, picture will be added once u/led

==3COM 3108 Wireless Phone==

[[File:3com_3108_1.jpg|300px|thumb|left|3COM 3108 Wireless Phone]]

'''Product:''' 3COM 3108 Wireless Phone

'''Company:''' 3COM

'''Overview:''' The 3Com 3108 Wireless Telephone is a Session Initiation Protocol (SIP)-based wireless Voice over Internet Protocol (VoIP) telephone. SIP is an internationally recognized standard <nowiki>(IETF RFC 3261)</nowiki> for implementing VoIP. You can make and receive VoIP calls as long as your Wireless Telephone is registered with a SIP proxy server and you are operating it within range of an IEEE 802.11b/g enabled wireless network (WLAN). The SIP proxy server can belong to a wireless Internet Telephony Service Provider (ITSP) or corporate VoIP PBX system, such as the 3Com NBX® System.

[[3COM_3108_Wireless_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Aastra 6730i/6731i VoIP Phone==

[[File:Aastra6730i.jpg|300px|frame|left|Aastra 6730i VoIP Phone]]

'''Product:''' Aastra 6730i/6731i VoIP Phone

'''Company:''' Aastra

'''Overview:'''The Aastra 6730i, a new member of the carrier-grade, open-standards based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.

Supported by a host of Aastra configuration options, XML development tools and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Atcom AG188N==

[[File:Atcom-ag188n.jpg|300px|thumb|left|Atcom AG188N]]

'''Product:''' Atcom AG188N

'''Company:''' Atcom

'''Overview:''' AG188N voice gateway is an Internet based one port voice gateway. AG188N ATA adapts multi voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.

AG188N ATA one FXS voice over IP gateway supports SIP, and IAX2 protocol, offering one 10/100Mbps Ethernet interface, one RJ11 telephone interface, one built-in router and lifeline port. AG-188N ATA is small and can be widely used in SOHO, small office and enterprise branches.

[[Atcom AG188N|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco Linksys PAP2==

[[File:Pap2.jpg|300px|thumb|left|Cisco Linksys PAP2]]

'''Product:''' Linksys PAP2

'''Company:''' Cisco

'''Overview:''' The Linksys PAP2 is an analog telephony adapter (ATA), which allows for the connection of up to two analog telephones to a softswitch or VoIP carrier using Session Initiation Protocol (SIP). The physical connections on the device are two RJ11 jacks for the telephones and one RJ45 jack for ethernet. The ethernet connection is 10 Mbit/s, half-duplex. The device is configured through a web interface.
Each analog telephone is presented as a distinct SIP user.

[[Cisco Linksys PAP2|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco Linksys PAP2T==

[[File:Pap2t.jpg|300px|thumb|left|Cisco Linksys PAP2T]]

'''Product:''' Linksys PAP2T

'''Company:''' Cisco

'''Overview:''' The Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports enables high-quality feature-rich VoIP service through your broadband Internet connection. Just plug it into your home router or gateway and use the two standard telephone ports to connect analog phones or fax machines. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and reliable fax connections, even while using the Internet.

With Internet telephony, along with low domestic and international phone rates, impressive arrays of special telephone features are available. Choose your preferred local dialing number, regardless of where you live. Or add a virtual telephone number to have forwarded to your Internet phone. You can even add a toll-free number. The Cisco PAP2T Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephone service provider, such as caller ID, call waiting, voicemail, call forwarding, distinctive ring, and much more.

[[Cisco Linksys PAP2T|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco Linksys SPA942 NA==

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

'''Overview:''' Four active lines and dual switched Ethernet ports offer incredible flexibility to the home or small-business Internet telephony user. Part of Cisco Small Business IP Phone Series, the SPA942 IP Phone can be configured to carry a unique phone number or extension for each of its lines, or can be configured to use a shared number over multiple phones.

Based on the SIP standard, the SPA942 is able to easily integrate new features and services offered by providers. A high-resolution display makes for easy menu- and web-based configuration.

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco IP Phone 7940/7960==

[[File:Cisco_7960g.png|300px|thumb|left|Cisco IP Phone 7940/7960]]

'''Product:''' Cisco IP Phone 7940/7960

'''Company:''' Cisco

'''Overview:''' The Cisco IP Phone 7940/7960 is a full-feature telephone that provides voice communication over an IP network. This phone functions like a traditional analog phone, allowing you to place and receive telephone calls. This phone also supports features, such as network call forwarding, redialing, speed dialing, transferring calls, placing conference calls, and accessing voice mail.

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco SPA2100 Phone Adapter==

[[File:Sipura2100.jpg|300px|thumb|left|Cisco SPA2100 Phone Adapter]]

'''Product:''' Cisco SPA2100 Phone Adapter

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco® SPA2100 Phone Adapter with Router (Figure 1) connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2100 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2100, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco SPA2100 Phone Adapter|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco SPA2102 Phone Adapter with Router==

[[File:Spa2102b.jpg|300px|thumb|left|Cisco SPA2102 Phone Adapter with Router]]

'''Product:''' Cisco SPA2102 Phone Adapter with Router

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco SPA2102 Phone Adapter with Router connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2102 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two 100BASE-T RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2102, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco_SPA2102_Phone_Adapter_with_Router|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco SPA504G Phone==

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

'''Overview:'''
The Cisco SPA504G uses standard encryption protocols to perform highly secure remote provisioning and
unobtrusive in-service software upgrades. Remote provisioning tools include detailed performance measurement
and troubleshooting features, enabling network providers to deliver high-quality support to their subscribers. Remote
provisioning also saves service providers the time and expense of managing, preloading, and reconfiguring
customer premises equipment.

[[Cisco_SPA504G_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream HandyTone 286==

[[File:Grandstream286.gif|300px|thumb|left|Grandstream HandyTone 286]]

'''Product:''' Grandstream HandyTone 286

'''Company:''' Grandstream

'''Overview:''' Grandstream's award-wining HandyTone-286 is innovative Analog Telephone Adaptor that offers a rich
set of functionality and superb sound quality at ultra-affordable price. They are fully compatible with SIP
industry standard and can interoperate with many other SIP compliant devices and software on the
market

[[Grandstream_HandyTone_286|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream HandyTone 486==

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

'''Overview:''' Grandstream's award-wining HandyTone-486 is an all-in-one VoIP integrated access device that features superb audio quality, rich functionalities, high level of integration, compactness and ultraaffordability. The HandyTone-486 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market.

Grandstream HandyTone-486 has been awarded the Best of Show product in 2004 Internet Telephony Conference and Expo.

[[Grandstream_HandyTone_486|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Grandstream HandyTone 502==

[[File:Ht502.jpg|300px|thumb|left|Grandstream HandyTone 502 - HT502]]

'''Product:''' Grandstream HandyTone 502 - HT502

'''Company:''' Grandstream

'''Overview:''' The HT502 is a powerful VoIP router. The product's inclusion of an integrated high performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features.

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_502_-_HT502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Netgear WGR615V==

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]

'''Product:''' WGR615V

'''Company:''' Netgear

'''Overview:''' The WGR615V was never sold to directly to the public when it was new but since the model is discontinued, lots of them are being sold.
It is a wireless (802.11 G) router that happens to have a built-in ATA.
It is possible to use just the ATA part.

[[Netgear WGR615V|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==OBi110==

[[File:OBi110-ATA.jpg|300px|thumb|left|OBi110]]

'''Product:''' OBi110

'''Company:''' OBIHAI Technology Inc

'''Overview:''' With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet.

The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to a traditional phone service. If you do not have a traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi110|See Configuration Details]]
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==Panasonic KX-TGP 550==

[[File:Pana550b.jpg|300px|thumb|left|Panasonic KX-TGP 550]]

'''Product:''' Panasonic KX-TGP 550

'''Company:''' Panasonic

'''Overview:''' Panasonic KX-TGP 550 SIP Cordless Phone System allows you to have up to eight (8) phonenumbers.You can set up in several ways: for example, you can set thephone number for each handset. Or you can group the handsets bygroup setting and restrict the incoming calls receivals to the specifichandsets. Handsets if you need them.

*Support for 3 simultaneous network conversations
*CODEC: G.711a-law / G.711μ-law / G.722(wideband) / G.729a / G.726(32K)
*DECT radio technology
*2.1" Large LCD with white backlight on cordless handset
*Up to 6 DECT cordless handsets*1
*Support for up to 8 SIP registrations (e.g. up to 8 DID lines or extensions)
*Hands-free speaker phone on cordless handset
*Wall mountable base unit

[[Panasonic KX-TGP 550|See Configuration Details]]
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==Pirelli DP-L10==

[[File:Pirelli1.jpg|300px|thumb|left|Pirelli DP-L10]]

'''Product:''' Pirelli DP-L10

'''Company:''' Pirelli

'''Overview:''' Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN 802.11b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet.

The Dual Phone enables mobile phone features such as Internet browsing, e-mail, SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings.

[[Pirelli DP-L10|See Configuration Details]]
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==Polycom SoundStation IP 4000 Conference Phone==

[[File:Polycom-4000-0.gif|300px|thumb|left|Polycom SoundStation IP 4000 Conference Phone]]

'''Product:''' Polycom SoundStation IP 4000 Conference Phone

'''Company:''' Polycom

'''Overview:''' the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the
benefits and versatility of a SIP enabled business. Designed for offices or small to medium sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated.

The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference.

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
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==Polycom SoundPoint IP 501, 550, 650, etc.==

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]

'''Product:''' Polycom SoundPoint IP 501, 550, 650, etc.

'''Company:''' Polycom

'''Overview:''' The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors’ IP solutions.

As protocols develop and standards evolve, it’s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment.

Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs.

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
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==Siemens Gigaset C450-Ip==

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|left|Siemens Gigaset C450-Ip]]

'''Product:''' Siemens Gigaset C450-Ip

'''Company:''' Siemens

'''Overview:''' The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.

[[Siemens Gigaset C450-Ip|See Configuration Details]]
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==Snom m3 VoIP Phone==

[[File:Snom_m3.png|300px|thumb|left|Snom m3 VoIP Phone]]

'''Product:''' Snom m3 VoIP Phone

'''Company:''' Snom

'''Overview:''' The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands free mode, calling line identification (CLI) by displaying name, number and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.

[[Snom_m3_VoIP_Phone|See Configuration Details]]
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==Telco AC-211==

'''Product:''' Telco AC-211

'''Company:''' Telco

'''Overview:''' Telco AC-211 is a SIP ATA supporting two lines and numerous configuration options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now defunct SunRocket service. This device works well with voip.ms once configured.

[[Telco_AC-211|See Configuration Details]]

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==Yealink SIP-T28P (VSRF)==

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink SIP-T28P

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.

[[Yealink_SIP-T28P|See Configuration Details]]
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==Zycoo ZP502==

[[File:ZP502.jpg|left|Zycoo ZP502]]

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

'''Overview:''' ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution,compatible with various Platforms such as Asterisk, FreePBX, Broadsoft, Cisco call manager etc.

[[Zycoo ZP502|See Configuration Details]]
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Telco AC-211

2011-06-17T01:28:51Z

Kage1: /* Configuration Details */

==Configuration Details==

'''Web Management'''

* Enter the IP in the address bar of your browser for the unit
* Enter the password for the unit and click Authencate

'''WAN Settings'''

* Go to '''WAN''' tab
* Select AutoConfiguration tab and uncheck "Enable Automatic Configuration"

'''SIP Settings'''

* Go to '''SIP''' tab
* '''Primary Server''': Enter the following.
**Enter the server name. Example: houston.voip.ms
**Port 5060
**Domain Name: voip.ms
**Check "Send Registration Request"
**Dial Plan: <nowiki>(>#|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|0|[489]11|*9[7-8]|*75xx)</nowiki>
**Transport: UDP
**User/Phone Num: Your SIP Username
**CallerID Name: Enter the CID you want to send
**Port*: 5060
**AEC On: On
**Authentication User Name: Your SIP Username
**Password: SIP Password
**Click '''Save SIP Settings'''

'''CODECS'''
* Go to '''CODECS''' tab
* Force preferred CODEC Line 1: Set your preferred CODEC, recommended is G711U
**The CODEC set must be enabled in your account: Account Settings -> Advanced
* Click '''Save CODEC Configuration'''

'''Miscellaneous'''
* Go to '''Miscellaneous''' tab
* Enter an NTP server
** Free NTP Servers available at http://www.pool.ntp.org/en/
**Click '''Save Settings'''

'''Reset'''
* Click the Reset button to reboot the gateway

==Notes==
* Under the SIP Server Settings the device supports a fail over server, but at this time the server does not support this. Leave this blank or you will have problems receiving incoming calls.
* The dial plan supports 10 & 11 digit dialing, *97 & *98 for voice mail, *75 dialing from the phonebook, 411, 811, & 911.
* When modifying the dialing plan >#| is required at the start of the plan, though I don't know why.

Telco AC-211

2011-06-17T00:59:48Z

Kage1: Created page with "==Configuration Details== '''Web Management''' * Enter the IP in the address bar of your browser for the unit * Enter the password for the unit and click Authencate '''WAN Se..."

==Configuration Details==

'''Web Management'''

* Enter the IP in the address bar of your browser for the unit
* Enter the password for the unit and click Authencate

'''WAN Settings'''

* Go to '''WAN''' tab
* Select AutoConfiguration tab and uncheck "Enable Automatic Configuration"

'''SIP Settings'''

* Go to '''SIP''' tab
* '''Primary Server''': Enter the following.
**Enter the server name. Example: houston.voip.ms
**Port 5060
**Domain Name: voip.ms
**Check "Send Registration Request"
**Dial Plan: <nowiki>(>#|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|0|[489]11|*9[7-8]|*75xx)</nowiki>
*** The dial plan supports 10 & 11 digit dialing. When modifying the dialing plan >#| is required, though I don't know why
**Transport: UDP
**User/Phone Num: Your SIP Username
**CallerID Name: Enter the CID you want to send
**Port*: 5060
**AEC On: On
**Authentication User Name: Your SIP Username
**Password: SIP Password
**Click '''Save SIP Settings'''

'''CODECS'''
* Go to '''CODECS''' tab
* Force preferred CODEC Line 1: Set your preferred CODEC, recommended is G711U
**The CODEC set must be enabled in your account: Account Settings -> Advanced
* Click '''Save CODEC Configuration'''

'''Miscellaneous'''
* Go to '''Miscellaneous''' tab
* Enter an NTP server
** Free NTP Servers available at http://www.pool.ntp.org/en/
**Click '''Save Settings'''

'''Reset'''
* Click the Reset button to reboot the gateway