Getting SIP to work to make calls to other cell phones with the ipod touch?
Currently I'm using Fring but so far I have only been able to receive calls and call other people who have the Fring application. Does anyone know how to make SIP calls using Fring application. Please help me.
You can try Fring or Skype to make calls .. Be sure to buy credits Learn how to make a free call via http://www.hackouriphone.com/2009/12/how-to-make-free-call-using-ipod-touch ipod touch. html
Since the mid-1990s IP telephony has become a widespread means of communication for businesses and service providers. Roughly three-quarters of large companies in the U.S. have already switched to IP telephony, enabling rich-media applications such as collaborative meetings, video, presence-based communication choices, rich hard phone or soft phone displays and end user call control mechanisms. Service provider backbone networks have also largely converted to VoIP transport realizing bandwidth and converged network architecture benefits. Yet TDM trunks are still the predominant mechanism to interconnect businesses with the PSTN (service provider), limiting the inter-business communications to the single-media (voice-only) transport of the traditional PSTN. To realize the promise of VoIP and enable rich-media business-to-business collaborative applications, service providers have in 2008 started offering implementable SIP trunk interconnects. Enterprise interest in SIP trunks for cost benefits, transport benefits as well as new productivity applications have also increased dramatically of late. SIP Trunks provides an overview of the trends and technologies in evolving PSTN interconnect from TDM to SIP-based transport. It discusses the real benefits and the popular myths surrounding SIP trunks and helps you evaluate what real benefits you could implement for your business. The book provides an in-depth discussion of planning your network for SIP trunk implementation and how to evaluate SIP trunk offerings. Practical guidance around RFP structure is given, including questions to ask the service provider, and providing a sample cost analysis. It also presents an in-depth discussion of how to deploy SIP trunk interconnects to an enterprise network. The possible deployment models are covered including the trade-offs and network design issues such as security considerations, call admission control and handling new call flows. It offers concrete implementation steps and realistic best practices to follow during the implementation. Network implementation is illustrated with a case study. The book concludes with an overview discussion of the future of unified communications networks and how business transformation due to end-to-end VoIP connectivity (of which SIP trunking is a critical piece) might evolve. This coverage helps the reader visualize how their business might transform over time and how best to plan and prepare their business to capitalize on the benefits.
“Aastra 6731i Brand New Includes One Year Warranty, Item # A6731-0131-10-01 The Aastra 6731i VOIP phone features a 3 line LCD display and supports up to 6 lines with call appearances. It offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms. The 6731i is well suited for daily telephone use in both small and large businesses. 6731i Features: 8 Programmable Line / Feature Keys, Corded Voice Over IP Phone, Voice Over Internet Protocol (VoIP), Session Initiation Protocol (SIP), Built-In HTTP Server, XML Browser, Up to 9 Call Appearance Lines, Caller ID / Call Waiting, Hearing Aid Compatible, Wall Mountable/Mount Included, Echo Cancellation, Speakerphone, 3-Line LCD Display, LED For Call & Message Waiting Indicator, 3-Way Conferencing, Speed Dial, Call Forwarding, Missed Call Indicator, Speakerphone Volume Control, Volume Control, Distinctive Ring Detection, Shared Call Appearances Allows You to Place a Call on Hold at One Set and Pick it up Easily at Another Set, Intercom, Call Transfer, Page, Auto Answer, Redial, Hold, Mute, 4 Navigational Keys, Do Not Disturb, Call Timer, Two-Port 10/100 Mbps Ethernet Switch, Headset Jack, Multilingual Menu Support (English, French, Spanish, Italian, & German), VoIP Features: Dual 10/100 Mbps Switched Ethernet Ports, Manual or DHCP IP Address Setup, Time / Date Synch Using SNTP, NAT, QOS, TOS, & Differentiated Services Code Point, Built-In HTTP/HTTPS Server For Web Administration & Maintenance, Voice Codec Support: G.711 -law / A-law, G.729, Protocols: IETF SIP (RFC3261 & Associated RFCs) The 6731i IP Phone requires the following environment: SIP-based IP PBX system or network installed and running with a SIP account created for the 6730i phone. Access to a Trivial File Transfer Protocol (TFTP), File Transfer Protocol (FTP), Hypertext Transfer Protocol (HTTP) server, or Hyper Text Transfer Protocol over Secure Sockets Layer (SSL) (HTTPS). 10/100 MB Ethernet/Fast Ethernet LAN, Category 5/5e Straight Through Cabling, Power over Ethernet (PoE) Inline Power Injector (Optional Accessory – Neces”
“Aastra MBU 400 Brand New Includes One Year Warranty, Item # A1762-0000-02-00 The Aastra MBU 400 SIP DECT 6.0 phone is an affordable, scalable multi-handset SIP DECT mobility solution for small business. The MBU 400 DECT gateway supports eight unique SIP registrations and one FXO port providing a fully integrated SIP and Analog service for small business. It can also be provisioned and managed via the AastraLink Pro 160 web UI, providing a single tool for configuration and maintenance of the handsets and other Aastra SIP terminals. Employing DECT 6.0 technology, the MBU 400 offers virtually interference-free performance in any work or residential environment. The handset is equipped with a 1.5″” backlit color graphical display providing caller ID, call waiting, DND, transfer, hold and a variety of information at-glance. This stylish handset also offers features such as speakerphone, intercom between handsets, polyphonic ring tones and a large directory for up to 170 entries. MBU 400 Features: DECT 6.0 Technology (1.9GHz) Interference-Free & Wide Range 30 more Battery life, Caller ID / Call Waiting, Expandable Up To 8 Handsets, Hearing Aid Compatible, Acoustic Echo Cancellation, Speakerphone, 1.5″” Full Color LCD Display 65k High Resolution, Adjustable Screen Contrast, Backlit Keypad, 30 Station Name / Number Caller ID Memory, 170 Station Phone Directory / Dialer, Call Forwarding, Missed Call Indicator, 9 Selectable Polyphonic Ringtones, Call Transfer, Redial, Hold, Date / Time Display, Do Not Disturb, Up to 12 Hours Talk Time, Up To 200 Hours Standby Time, Low Battery / Out of Range Indicator, Headset Jack, Belt Clip – Included w/ Brand New Models, Multi-Language Menus, Base Unit Features: New DECT 6.0 Technology -1.9GHz SIP DECT Cordless Base, 1 FXO Port & 1 LAN Port, Multiple Handset Capability: – Support For 8 Paired 420d DECT Handsets Each w/ A Unique VoIP Telephone Number, Centralized/Shared Phonebook For All Paired Handsets, Embedded HTTP Web-server For Configuration, Integrated Antennas, 50m Indoor & 300m Outdoor Range Additional Handsets For This Model Are: span class=”
“Aastra 6731i w/ AC Adapter Brand New Includes One Year Warranty, Item # A6731-0131-10-02 The Aastra 6731i VOIP phone features a 3 line LCD display and supports up to 6 lines with call appearances. It offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms. The 6731i is well suited for daily telephone use in both small and large businesses. 6731i w/AC Adapter Features: 8 Programmable Line / Feature Keys, Corded Voice Over IP Phone, Voice Over Internet Protocol (VoIP), Session Initiation Protocol (SIP), Built-In HTTP Server, XML Browser, Up to 9 Call Appearance Lines, Caller ID / Call Waiting, Hearing Aid Compatible, Wall Mountable/Mount Included, Echo Cancellation, Speakerphone, 3-Line LCD Display, LED For Call & Message Waiting Indicator, 3-Way Conferencing, Speed Dial, Call Forwarding, Missed Call Indicator, Speakerphone Volume Control, Volume Control, Distinctive Ring Detection, Shared Call Appearances Allows You to Place a Call on Hold at One Set and Pick it up Easily at Another Set, Intercom, Call Transfer, Page, Auto Answer, Redial, Hold, Mute, 4 Navigational Keys, Do Not Disturb, Call Timer, AC Powered (Adapter Included), Two-Port 10/100 Mbps Ethernet Switch, Headset Jack, Multilingual Menu Support (English, French, Spanish, Italian, & German), VoIP Features: Dual 10/100 Mbps Switched Ethernet Ports, Manual or DHCP IP Address Setup, Time / Date Synch Using SNTP, NAT, QOS, TOS, & Differentiated Services Code Point, Built-In HTTP/HTTPS Server For Web Administration & Maintenance, Voice Codec Support: G.711 -law / A-law, G.729, Protocols: IETF SIP (RFC3261 & Associated RFCs) The 6731i IP Phone requires the following environment: SIP-based IP PBX system or network installed and running with a SIP account created for the 6730i phone. Access to a Trivial File Transfer Protocol (TFTP), File Transfer Protocol (FTP), Hypertext Transfer Protocol (HTTP) server, or Hyper Text Transfer Protocol over Secure Sockets Layer (SSL) (HTTPS). 10/100 MB Ethernet/Fast Ethernet LAN, Category 5/5e Straight Through Cabling, Power over Ethe”
This book explains why people and companies are using SIP equipment and software to efficiently upgrade existing telephone systems, develop their own advanced communications services, and to more easily integrate telephone network with company information systems. This book provides descriptions of the function parts of SIP systems along with the fundamentals of how SIP systems operate. It identifies and explains what key services are possible through the use of SIP and how existing phone systems can be upgraded to SIP capabilities. It describes why it is easy to integrate SIP with information systems along with how to develop new advanced revenue producing services. The reader will learn the basic SIP system development and installation process and how to manage SIP systems. Explained are the typical costs of SIP systems and how SIP technology is changing to meet future multimedia communication needs.
“Aastra 9143i / 33i Brand New Includes One Year Warranty, Item # A1733-0131-10-05 The Aastra 9143i is a SIP based voice over IP telephone has excellent XML software capabilities and comes with a solid hardware feature set. This IP phone includes dual PoE compatible ethernet ports and a power supply. The headset port is modular and amplified and the LCD display is backlit. The 9143i will appeal to those requiring advanced SIP features in a traditional phone design. 9143i Features: Call Waiting, Corded Voice Over IP Phone, Voice Over Internet Protocol (VoIP), Session Initiation Protocol (SIP), Built-In HTTP Server, XML Browser, Up to 9 Call Appearance Lines, Modular Headset Connector With Built-In Amplifier, Hearing Aid Compatible, Echo Cancellation, Full Duplex Handset SpeakerPhone, 3-Line Backlit LCD Display, LED For Call & Message Waiting Indicator, Phone Directory / Dialer, 3-Way Conferencing, 7 Programmable Memory Dial Buttons, Call Forwarding, Missed Call Indicator, Adjustable Receiver, Speakerphone, Headset Volume Controls, Distinctive Ring Detection, Shared Call Appearances Allows You to Place a Call on Hold at One Set and Pick it up Easily at Another Set, Intercom, Call Transfer, Redial, Mute, Live Dialpad Option, 4 Navigational Keys, Do Not Disturb, Call Timer, Lamp Flasher Jack Lamp Button Operates Connected Light Source, Two-Port 10/100 Mbps Ethernet Switch, Multilingual Menu Support (English, French, Spanish, Italian, & German), Caller and Calling Line Information Click Below to View Additional Details Requirements: SIP-based IP PBX System or Network Installed & Running w/ a SIP Account Created For the 9143i IP Phone. Access to a Trivial File Transfer Protocol (TFTP), File Transfer Protocol (FTP), Hypertext Transfer Protocol (HTTP) server, or HyperText Transfer Protocol over Secure Sockets Layer (SSL) (HTTPS). Ethernet/Fast Ethernet LAN (10/100 Mb), Category 5/5e Straight Through Cabling Security: User & Ad”
“Aastra 9480i / 35i Brand New Includes One Year Warranty, Item # A1735-0131-10-05 The Aastra 9480i is a SIP based voice over IP telephone designed for the executive. The Aastra 9480i has excellent XML software capabilities and comes with a solid hardware feature set. This phone includes dual PoE compatible ethernet ports & a power supply. The LCD display is backlit and the SIP support is exceptional. Part of the Aastra IP Series of telephones, the 9480i is ideal for moderate to heavy telephone users who require more one touch feature keys and XML based programs. 9480i Features: Corded Voice Over IP Phone, Voice Over Internet Protocol (VoIP), Session Initiation Protocol (SIP), Built-In HTTP Server, XML Browser, Up to 9 Call Appearance Lines, Caller ID / Call Waiting, Modular Headset Connector With Built-In Amplifier, Hearing Aid Compatible, Echo Cancellation, Full Duplex Handset SpeakerPhone, Backlit LCD Display, LED For Call & Message Waiting Indicator, Phone Directory / Dialer, 3-Way Conferencing, Call Forwarding, Missed Call Indicator, Adjustable Receiver, Speakerphone, Headset Volume Controls, Intercom, Call Transfer, Page, Redial, Hold, Mute, Pause, Live Dialpad Option, 4 Navigational Keys, Do Not Disturb, Call Timer, Lamp Flasher Jack Lamp Button Operates Connected Light Source, Two-Port 10/100 Mbps Ethernet Switch, Multilingual Menu Support (English, French, Spanish, Italian, & German), Click Below to View Additional Details Requirements: SIP-based IP PBX system or network installed and running with a SIP account created for the 9480i phone. Access to a Trivial File Transfer Protocol (TFTP), File Transfer Protocol (FTP), HypertextTransfer Protocol (HTTP) server, or Hyper Text Transfer Protocol over Secure Sockets Layer (SSL) (HTTPS). Ethernet/Fast Ethernet LAN (10/100 Mb), Category 5/5e straight through cabling Security: User & Administrator Level Passwords For Login, Encryption of Configuration Files, HTTPS Confi”
“Aastra DECT 142 Brand New Includes One Year Warranty, Item # D0068-895D-00-00 The Aastra DECT 142 SIP handset features DECT technology that offers superior levels of interference-free performance, security and reliability. DECT 6.0 technology virtually eliminates issues of dropped calls, interference, security and range by employing IP DECT access points that seamlessly “”handoffs”" calls and enables roaming. DECT 142 Features: Worldwide Calling Sign Up, Microphone Mute, Signal Strength Indicator, New DECT 6.0 Technology – Interference Free Communication – Increased Clarity / Enhanced Security – Wider Range / Network Friendly, 1.9GHz Extra Handset / Charger – Dedicated for Cordless Phones by the FCC – 30 Battery Life Increase Over 5.8GHz, Up To 300, 000ft Range, Caller ID / Call Waiting, Speakerphone, 4-Line, 16-Character Backlit Display, Adjustable Screen Contrast, Illuminated Keypad, LED Status Indicator, On-Hook Dialing, 100 Station Phone Directory / Dialer, Call Block, Speed Dial, Call Forwarding, Message Waiting Indicator, Missed Call Indicator, Earpiece / Ringer Volume Control, Call Transfer, Auto Answer, Ringer Off Option, Vibrating Ringer, Redial, Hold, Date / Time Display, Do Not Disturb, Alarm, Low Battery Indicator, 2.5mm Headset Jack, Keypad Lock, Multi-Language Menus, Will Not Work Without The Aastra RFP 32 IP DECT Access Point”
- Snom VoIP wireless DECT phone- 3 Concurrent VoIP calls (SIP 2.0)- Up to 8 handsets- Call transfer (attended and blind); call hold SIP- Call log: incoming, outgoing and missed calls- Call swap- Intercom – handset to handset via gateway- Phonebook 170 ent
State-of-the-art SIP primer SIP (Session Initiation Protocol) is the open standard that will make IP telephony an irresistible force in communications, doing for converged services what http does for the Web. "SIP Demystified" – authored by Gonzalo Camarillo, one of the contributors to SIP development in the IETF–gives you the tools to keep your company and career competitive. This guide tells you why the standard is needed, what architectures it supports, and how it interacts with other protocols. As a bonus, you even get a context-setting background in data networking. Perfect if you’re moving from switched voice into a data networking environment, here’s everything you need to understand: * Where, why, and how SIP is used * What SIP can do and deliver * SIP’s fit with other standards and systems * How to plan implementations of SIP-enabled services * How to size up and choose from available SIP products
