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<span style="font-size:400%; line-height: 1.3em;"><center>VoIP.ms</center></span>

<div style="text-align:center; font-family:Arial; word-spacing: 60px"><ins>'''[[Bienvenue|Français]]'''</ins> <ins>'''[[Bienvenido|Español]]'''</ins> </div>

__TOC__

<table style="font-family:Arial; width:60%; border: 1px solid black; border-collapse:collapse" align="center">
<tr style="background-color:darkgray; color:white">
<th><ins>'''Popular Articles'''</ins> </th>
<th><ins>'''Frequently Asked Questions'''</ins> </th>
<th><ins>'''Popular Devices'''</ins></th>
</tr>
<tr>
<td>[[Getting Started|Getting Started]] </td>
<td>[[FAQ#Can_I_port_my_existing_number_from_another_provider_to_VoIP.ms.3F|Can I Port My Number...?]] </td>
<td>[[Cisco_SPA112|Cisco SPA112]]</td>
</tr>
<tr>
<td>[[Voicemail|Voicemail]] </td>
<td>[[FAQ#Can_I_use_my_existing_device_with_VoIP.ms.3F|Can I Use My Existing device with VoIP.ms?]]</td>
<td>[[OBi_100/110|OBi 100/110]]</td>
</tr>
<tr>
<td>[[Choosing Server|Choosing Server]] </td>
<td> [[FAQ#Can_I_register_2_or_more_different_devices_with_the_same_account_username.3F|Can I Register 2 Different Devices...?]] </td>
<td>[[Grandstream_HandyTone_702_-_HT702|Grandstream HandyTone 702]]</td>
</tr>
<tr >
<td>[[Manage DID|Manage DID]] </td>
<td>[[Choosing_Server#IPs|What are the IP addresses of VoIP.ms' servers ?]]</td>
<td>[[Cisco_Linksys_PAP2T|Cisco Linksys PAP2T]]</td>
</tr>
<tr>
<td><span style="color:blue">More in The Guides Section</span> </td>
<td>[[FAQ|See all>]] </td>
<td> [[Devices|See all>]] </td>
</tr>
</table>

== VoIP.ms Wiki ==

Welcome to the VoIP.ms Community Wiki! VoIP.ms is the most reliable, affordable and customizable VoIP experience on the market and this Wiki will help navigate the features.

In this wiki you can find guides, information and configuration examples for configuring your PBX, Device or Softphone.

We encourage users from the community to collaborate along with VoIP.ms staff to add guides, information and other articles pertinent to the service and products.

If you want to help by contributing with a new article or adding information to existing articles, please write to [mailto:support@voip.ms support@voip.ms] requesting for edit permission for the username you use in this wiki.

==Configuration Samples==
<span style="font-size:200%; line-height: 1.3em;"><center>[[PBXs]] - [[Devices]] - [[Softphones]]</center></span>

==Contact Us==

* [https://livechat.boldchat.com/aid/2947277729005480016/bc.chat?cwdid=60236691424546376&url=https%3A//www.voip.ms/ Live Chat Support]
* Technical Support - [mailto:support@voip.ms support@voip.ms]
* Feedback - [mailto:feedback@voip.ms feedback@voip.ms]
* Porting - [mailto:ports@voip.ms ports@voip.ms]
* Sales and General Information - [mailto:sales@voip.ms sales@voip.ms]
** Toll Free, USA/Canada - 1.877.7.VOIP.MS
** Worldwide - 1.214.615.8599
* FAX
** Toll Free, USA - 1.888.311.7782
** Dallas, Texas, USA - 1.214.723.7555

==Social Media==

* [https://www.facebook.com/VoIP.ms Facebook]
* [https://www.facebook.com/VoIP.ms Twitter]

2013-03-15T16:06:52Z

FherNando: /* OBi110 */

==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]]<br>
[[PAP2T With Static IP|See Configuration Details with Static IP]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco SPA112==

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]

'''Product:''' SPA112

'''Company:''' Cisco

'''Overview:''' The Cisco SPA112 2 Port Adapter enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection. Easy to install and use, it works over an IP network to connect analog phones and fax machines to a VoIP service provider and provides support for additional LAN connections.

The Cisco SPA112 includes two standard telephone ports to connect existing analog phones or fax machines to a VoIP service provider. Each phone line can be configured independently. With the Cisco SPA112, users can protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines as well as control their migration to IP voice with an extremely affordable, reliable solution.

[[Cisco SPA112|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. Phones require Power Over Ethernet (PoE) or [http://www.ciscopowercube.com Cisco CP-PWR-CUBE] 48V AC Adapter to power up.

[[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>

==Mediatrix 4100 Series==

[[File:Mediatrix4102s.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix 4102S and 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® 4100 Series device is a security ready VoIP gateway, connecting up to two analog phones and/or faxes, as well as a PC or a home router to a broadband modem. The Mediatrix 4102 offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling and media transmission aspects.

[[Media5 Mediatrix 4100|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:''' OBi 100/110

'''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.

[[OBi 100/110|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

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

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

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

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

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

==SNOM 320 VoIP Phone==

[[File:Snom320.png|300px|frame|left|Snom 320 VoIP Phone]]

'''Product:''' Snom 320 VoIP Phone

'''Company:''' Snom

'''Overview:'''Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with built-in, full-duplex speakerphone and three-party conference bridging.

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

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

==Telco AC-211==

[[File:Ac211n.jpg|300px|thumb|left|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]]

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

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

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

OBi 100/110

2013-03-15T16:04:33Z

FherNando: Undo revision 5308 by FherNando (talk)

[[File:OBi110-ATA.jpg|200px]]

The Obi100 is a 1-phone port ATA adapter that support SIP VoIP services, the OBi100 is perfect for customers who do not have a traditional phone service, yet need similar solution and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your Obi100 you will need to plug it to your router/modem via its Internet port with an Ethernet cable and connect a regular handset phone to its Phone port, then follow the next steps.

== Manual Configuration details ==

'''**NOTE :''' You may use this guide to configure an Obi110 as well.

Dial '''* * * 1''' from the connected phone, to get the IP address of your device. This will be a number similar to 192.168.X.X
Once you get the IP address, enter it in the URL address bar (http://) of your Internet Browser to get access to the Graphic User Interface of the Obi100.
If done properly, the following window should appear on your screen:

[[File:ObiLogin.png]]

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the Obi interface.
Now on the left side of the screen please find the next options and follow the next steps:

'''Service Providers >> ITSP Profile A >> SIP'''

In this section you can set the server and the port that you wish to register to.

[[File:ObiITSP.png]]

Please note that in order to change the settings, you need to Uncheck the Default box on the right hand side.

*ProxyServer: atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location.)
*ProxyServerPort: 5060
*RegistrarServer: atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location.)
*RegistrarServerPort: 5060

'''Service Providers >> ITSP Profile A >> General'''

In this section you can set the Name, protocol and DTMF method to connect through.

[[File:ObiITSPGeneral.png]]

*Name: 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Voice Services >> SP1 Service'''

In this section you can set your Main account/sub_account credentials like User name and Password. The Main account password by default is the same password as the Customer portal.

[[File:ObiSP1.png]]

*AuthUserName: 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
*AuthPassword: ********* (SIP Account Password)

'''Checking your Voicemail'''

If you encounter any issues when you try to check your VoiP.MS [[Voicemail]]. Please try the following suggestions.

Try changing your Dial plan.
Replace the 555 digits in the following lines by the area code of your choice and copy the line, including parenthesis, in the Digitmap field in the ITSP Profile:

Default Dial Plan:
((1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*]@@.)

Dial Plan to allow you to dial other numbers plus those beginning with * followed by one or more numbers:
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|(Mipd)|[^*]@@.)

Dial Plan to allow you to dial other number plus *97 and *98 ONLY:

(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*9[78]|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

''' ''An additional note regarding Outbound Calling'''''

In at least one instance it was necessary to specify a non-default outbound calling route in the Obi110 to be able to place calls using the voip.ms service. The default setting had the Obi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG]]

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

[[Category:Analog Telephone Adapters]]

== Configuration Using OBi Dashboard ==

I had added my device to the OBitalk network prior to trying the manual configuration details given on this page. I then attempted the manual configuration explained above. I found that after I submitted the changes, the settings would revert back to the default values after a short amount of time (I suspect a configuration mismatch was detected by the OBiTalk network and the previous configuration was automatically restored, but this is just supposition on my part). By configuring the device through the OBitalk dashboard and selecting ''voip.ms'' as the service provider the device was up and working with little effort.

Add your device to the OBitalk service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the obi110 and are not discussed here.

After the obi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your obi110.

== 10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98 ==

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.

OBi 100/110

2013-03-15T15:59:31Z

FherNando: Reverted edits by Neocypher (talk) to last revision by Alan

[[File:OBi110-ATA.jpg|200px]]

The Obi100 is a 1-phone port ATA adapter that support SIP VoIP services, the OBi100 is perfect for customers who do not have a traditional phone service and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your Obi100 you will need to plug it to your router/modem and connect a regular handset and follow the next steps.

== Manual Configuration details ==

Dial ***1 from the connected phone, to get the IP address of your device.
Once you get the IP address, enter it in the address bar of your Internet Browser to get access to the Graphic User Interface.

[[File:ObiLogin.png]]

Once you could see the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

When you had entered this you will be able to see the Obi interface, now at the left side of the screen please find the next options and follow the next steps:

'''Service Providers >> ITSP Profile A >> SIP'''

In this section you could set the server and the port that you wish to register to.

[[File:ObiITSP.png]]

Please note that in order to change the settings, you need to Uncheck the Default box on the right hand side.

*ProxyServer: atlanta.voip.ms (one of our multiple servers)
*ProxyServerPort: 5060
*RegistrarServer: atlanta.voip.ms (one of our multiple servers)
*RegistrarServerPort: 5060

'''Service Providers >> ITSP Profile A >> General'''

In this section you could set the Name, protocol and DTMF method to connect through.

[[File:ObiITSPGeneral.png]]

*Name: 100000 (Replace with your 6 digits UserID or sub account user name, i.e. 123456 or 123456_sub)

'''Voice Services >> SP1 Service'''

In this section you could set your account/sub_account credentials as User name and Password. Any time you wish to confirm your voip.ms credentials you could refer to the next link: https://www.voip.ms/m/samples/voxalot.php

[[File:ObiSP1.png]]

*AuthUserName: 100000 (Replace with your 6 digits UserID or sub account user name, i.e. 123456 or 123456_sub)
*AuthPassword: ********* (Account Password)

'''Note:'''
If your VoIP.ms has sub-account(s) you may need to enter that number, e.g. 100000_1, for Name and AuthUserName.

'''Checking your Voicemail'''

If you encounter any issues when you try to check your VoiP.MS [[Voicemail]]. Please try the following suggestions.

Try changing your Dial plan.
Replace the 555 digits in the following lines by the area code of your choice and copy the line, including parenthesis, in the Digitmap field in the ITSP Profile:

Default Dial Plan:
((1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*]@@.)

Dial Plan to allow you to dial other numbers plus those beginning with * followed by one or more numbers:
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|(Mipd)|[^*]@@.)

Dial Plan to allow you to dial other number plus *97 and *98 ONLY:

(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*9[78]|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

''' ''An additional note regarding Outbound Calling'''''

In at least one instance it was necessary to specify a non-default outbound calling route in the Obi110 to be able to place calls using the voip.ms service. The default setting had the Obi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG]]

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

[[Category:Analog Telephone Adapters]]

== Configuration Using OBi Dashboard ==

I had added my device to the OBitalk network prior to trying the manual configuration details given on this page. I then attempted the manual configuration explained above. I found that after I submitted the changes, the settings would revert back to the default values after a short amount of time (I suspect a configuration mismatch was detected by the OBiTalk network and the previous configuration was automatically restored, but this is just supposition on my part). By configuring the device through the OBitalk dashboard and selecting ''voip.ms'' as the service provider the device was up and working with little effort.

Add your device to the OBitalk service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the obi110 and are not discussed here.

After the obi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your obi110.

== 10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98 ==

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.