November 2015

• 09 – 10 OpenSIPS Summit 2015 in Austin, TX - Come and help us "Keep Austin Wierd" at our eclectic venue, The Contemporary Austin Jones Center. We'll be presenting, via presentations and workshops, all the great additions we have in OpenSIPS 2.1 - webRTC, async queries, SIP compression, fraud detection and many others. The Summit is not only about 2.1, but also about tools, platforms or engines which are OpenSIPS interconnected (like billing, interfaces and others). We'll also have presentations from companies using OpenSIPS in their production environments, giving us insight into it's real world applications.
• 11 – 13 OpenSIPS LIVE BOOTCAMP 2015 in Austin, TX - Enjoy a rare opportunity to train with the founder of the OpenSIPS Project Bogdan-Andrei Iancu! As part of OpenSIPS week in Austin, we'll be holding a LIVE training following the conclusion of the OpenSIPS Summit on Nov 10th. If you've ever wanted to become an OpenSIPS Certified Professional, this is you're chance! You'll have the rare opportunity to train live with the founder of the OpenSIPS Project Bogdan-Andrei Iancu! It's not a secret that there is a growing demand for OpenSIPS Certified Professionals all over the world. This training is intended to offer you a 50/50 view between theoretical and practical applications of OpenSIPS. It covers advanced topics that are critical to telecom network operators in regards to security, scaling, and integration. For participants that would like an introduction to OpenSIPS basics and to qualify for prerequisites to the advanced topics, there will be a 2 day online primer prior to the live training.

October 2015

• 06 – 08 ITEXPO Anaheim 2015 - ITEXPO brings together service providers, enterprises, government agencies, resellers, vendors and developers to discuss the latest innovations that are changing the marketplace. Conference topics include WebRTC, cloud computing, unified communications, software defined networking (SDN), network functions virtualization (NFV), channel strategies and more. ITEXPO will be held October 6 - 8, 2015, at the Anaheim Convention Center in Anaheim, California.
• 18 - 22 GITEX 2015 - 35th annual GITEX Technology Week in Dubai, UAE, 18-22 October. Speedflow will be among the recognizable exhibitors at the show, which acts as a hub for the Middle East, Africa and South Asia technologies industry.

September 2015

• 28 – 29 INDO AFRICA ICT EXPO 2015 - At Indo - Africa ICT Expo 2015, the latest innovative ICT technologies covering wide range of products, services, and software applications will be on display and open for technology transfer. ICT Expo scheduled at Nairobi, Kenya.
• 16 – 18 Cloud Partners Conference - Cloud Partners Conference is a good platform to discover the cloud based solutions that can scale with your business needs. Cloud Partners Conference scheduled at Hynes Convention Center, Boston.
• 01 – 04 Asian Carriers Conference 2015 (ACC) - The best and the biggest information technology and telecommunications companies from Asia and the rest of the world. ...

How Does Mobile VoIP Work?

• Anytime minutes
• Night or weekend minutes
• Rollover minutes
• Roaming charges
• Incoming call charges
• Messaging limits
• Mobile-to-mobile calling (check with your mobile VoIP provider, some do treat in-network calls differently)

Accessing Mobile VoIP

SIP Phones

• 67xxi Wallmount Bracket - Wall mount bracket which allows for simple and efficient installation of multiple models of Aastra IP phones.

• 6735i - 5 line text display - 6 softkeys, 6 programmable keys, gigabit ethernet
• 6737i - Dual 5 line text display, 12 softkeys, gigabit ethernet
• 6739i - Color graphical display, resistive touch screen

• 6757i - Dual 5 line text display, 12 softkeys
• 6757i CT - Includes cordless handset (North America / Australia / New Zealand only)
• 6755i - 5 line text display - 6 softkeys, 6 programmable keys
• 6753i - 3 line text display - programmable buttons
• 6751i - lobby phone

• 6731i **New entry level phone
• 6730i Similar to 6731i except no support for PoE and only 1 ethernet port.

• 9480i - 5 line text display, 6 softkeys
• 9480i CT - Includes cordless handset (North America / Australia / New Zealand only)
• 9143i - 3 line text display, programmable keys

Expansion Modules

• M670i (previously 536M) - For the 53i, 55i, 57i, 57i CT and 39i. Provides 36 additional keys with LED status display. ...

What is SBO?

Why Bandwidth Saver is so important for VoIP?

How does SBO work?

Key Features of SBO:

• The most important feature of SBO is, it can reduce bandwidth cost directly by 80% without degrading service quality.
• SBO works behind any type of firewall and NAT. That means, it has anti-block feature.
• Works with all commonly used codecs such as G.729, G.723 etc.
• Multipath: SBO Multipath allows you to use multiple number of internet connections same time. These connections can be used simultaneously for your VoIP termination establishment which can balance load among available networks and it develop service quality expectedly.
• Works with any type of internet connection i.e. GPRS, EDGE, 3G, 4G, Wi-Fi, Wi-Max, so it is possible to setup anywhere where mobile internet is available.
• Highly secure and linux based software.
• Full featured billing platform

• SBO
• 3CX
• 4PSA
• A2Billing Softswitch & VoIP Billing Software
• akant
• All Call Technologies @allcalltech
• Aptus FonB
• Asterisk Solution
• AstraQom Canada
• Askozia
• Astcon - Danish one-stop-shop ASTCON.DK
• asterisknow
• asteriskpbx
• astricon
• beelinebill Bill Miller, Digium
• biznews_voip
• Business Phone Systems
• ToIP Opensource Integrator , Lausanne, Switerland
• call_accounting Call Accounting Mate at www.callaccounting.ws
• CallnFax CallnFax delivers VoIP and Faxing, best rates and service: CallnFax.com
• CbeyondInc
• codewhiz Coder, Asterisk hacker, @digium reseller(?)

Welcome to the VOIP Wiki - a reference guide to all things VOIP.

NEWS

• 2015-11-03 - Thirdlane advances scalability and adds new features to its Business, Multi Tenant, and Elastic Cloud PBX platforms
• 2015-11-3 -What types of Internet Connections are Supported by SBO
• 2015-11-02 - OpenSIPS Summit 2015 in Austin, TX
• 2015-10-30 - Multi-Carrier Management – Valuable Feature to Look for in a Softswitch
• 2015-10-26 - Kamailio Advanced Training, during Nov 30-Dec 02, 2015, in Berlin, Germany
• 2015-10-22 -QueueMetrics 15.10 enriches Asterisk call-centre reporting with skills, tags and features
• 2015-10-22 - Yeastar Releases New 1 FXS and 2 FXS Analog Telephone Adapter TA100 & TA200
• 2015-10-20 - OpenSIPS Summit is set, meet onSIP, Twilio, Jibe, Homer, SippySoft, Bandwidth
• 2015-10-19 - Hosted PBX – The Cloud’s Coolness Factor
• 2015-10-15 - Bicom Systems is excited to release our first FREE fact sheet with two secrets to earning more profit
• 2015-10-15 - MediaCore SMS - updated cost-effective carrier SMS solution
• 2015-10-15 - End-point Options Available for VoIP Call Shop
• 2015-10-14 - New version astercc gives you no answer call report based on early media.
• 2015-10-11 - The Innitel Telecom network keeps on expanding.
• 2015-10-07 -

November 2015

• 2015-11-03 - Thirdlane releases new version 7.4 of its Business, Multi Tenant, and Elastic Cloud PBX platforms

October 2015

• 2015-10-22 - QueueMetrics 15.10 released - New and updated features
• 2015-10-19 - miniSIPServer V23 is released to support "shared appearance" feature.
• 2015-10-14 -

Hard Phones

General

• Audiocodes HD IP Phones
• 2 Day Direct Top of the line VoIP phones
• 3Com - complete line of SIP business phones

• China SIP Trunking (CTS): A Leading Business VOIP Solutions Provider in China. Hosted PBX in China, SIP Trunking in China, DID Number, Toll Free Number, VOIP Consulting, VOIP System Maintenance, VOIP Architect, VOIP Specialist, Microsoft Lync Integration
• China Hosted PBX (CTS): A Leading Business VOIP Solutions Provider in China. Hosted PBX in China, SIP Trunking in China, DID Number, Toll Free Number, VOIP Consulting, VOIP System Maintenance, VOIP Architect, VOIP Specialist, Microsoft Lync Integration
• BOL - VoIP Service Provider's Provider. Provides both H.323 and SIP softswitch, wholesale system, prepaid/postpaid realtime RADIUS billing, customizable softphone (encryption function included), softphone SDK, calling card, callback system and SIP Session Border Controller with encryption function. VoIP services: Wholesale(86), H323/SIP A-Z Route, Co-location at HongKong.
• DIDWW:Chinese DID available now instantly on on-line store.
• http://www.eztor.com-professional China VOIP provider.with high quality and low cost.
• International Carrier Exchange, Room 701, Oversea Chinese, Venture Building, Shenzhen, China
• Live-TechOffers TalkSwitch VoIP Hardware & Lynk VoIP Services | Remote Technical Support | VoIP Service Plans | Worldwide Support
• Mydivert.com China DID Provider offers local VoIP DID Virtual Phone Numbers. A low cost virtual international telephone number with free incoming calls and call forwarding
• http://www.vidanetwork.net/ip-pbx-bundle-en.php Reseller needed. Provides Business IP-PBX, ATA, IP-phone, DID numbers and service bundles in USA, Canada, China and India.
• http://www.voipcallsecure.com VoIPCall Secure provides subscribers secure-encrypted phone service to businesses and high net worth individuals who want to keep their private phone calls private. Service available across the globe. Includes unlimited subscriber to subscriber calling and pay for service international calling.
• http://www.voptech.comVoptech is a professional VoIP products provider,we have life long technical support,our products both competitive in price and quality.We also had been passed the Elastix compatible test already.IPLAN Argentine (http://iplan.com.ar/), a leading VoIP service provider in Latin America, orders several thousands of IP Phone VI2006 from us every month. We have VoIP Gateway with up to 96 FXS/FXO Ports. It had been certified by Mitel Canada (www.mitel.com), together with Cisco, AudioCodecs Gateways.

Asterisk sip.conf, peer definition: canreinvite option

Versions

Description

• If one of the clients is configured with canreinvite=NO, Asterisk will not issue a re-invite at all.
• If the clients use different codecs, Asterisk will not issue a re-invite.
• If the Dial() command contains ''t'', ''T", "h", "H", "w", "W" or "L" (with multiple arguments) Asterisk will not issue a re-invite.

• canreinvite = yes "allow RTP media direct"
• canreinvite = no "deny re-invites"
• canreinvite = nonat "allow reinvite when local, deny reinvite when NAT"
• canreinvite = update "use UPDATE instead of INVITE"
• canreinvite = update,nonat "use UPDATE when local, deny when NAT"

Notes

• reinvite=yes/no is plain wrong, even if you see it mentioned in example .conf files. The correct syntax is canreinvite=yes/no
• Connecting media paths direct to an endpoint behind NAT won't be pretty. Especially if both devices are behind NAT. You might want to try using SER's nathelper in conjunction since nathelper.so can rewrite the SDP so that the private IP addresses are not included in the re-invite.
• When dtmfmode=rfc2833, asterisk will send the RTP stream through asterisk. With dtmfmode=info canreinvite works properly.
• Asterisk 1.8 added the media_address= configuration option which can be used to explicitly specify the IP address to use in the SDP for media (audio, video, and text) streams.

Background info

ConfBridge

Synopsis

Description

10.x

Asterisk cmd ConfBridge

• 'a' — 設為管理員
• 'A' — 設為標記成員
• 'c' — 計算有多少匿名成員進入會議室
• 'm' — 設定一開始就為靜音模式
• 'M' — 啟用音樂等候，當會議室只有一人時，可特別指定音樂等候的等級，若未指定則使用目前頻道的設定或預設值
• '1' — 不撥放提示訊息，當第1位使用者進入時
• 's' — 當收到 # 號鍵時，出現語音選單 (普通成員是普通選單、管理員是管理選單)
• 'w' — 等待標記成員進入後會議才開始(若有啟用 M 參數，則在那之前都會一直撥放等候音樂)
• 'q' — 靜音模式 (不撥放進入/離開訊息)

Muteing

不同於 MeetMe

Asterisk consultants: Greece

Modulus SA

• Home page:http://www.modulus.gr
• Telephone: +30-215-215-1500
• Email: telecoms at modulus.gr

Netvoice.gr

• Home page:http://www.netvoice.gr
• Home page:http://www.netvoice.eu
• Telephone: +30-211-800-3222
• Email:info@netvoice.gr

VOIPsystems.gr

• Home page:http://www.voipsystems.gr
• Telephone: +30-2310-460660
• Email:info@voipsystems.gr

MerIT Advanced IT Solutions, Athens

• Home page:: http://www.mer-it.gr
• Telephone: +30-2109231296
• Email:info@mer-it.gr

Transcom Ltd., Athens

• Home page:: http://www.transcom.gr
• Telephone: +30-2106564630
• Email:info@transcom.gr

SEQUENCE ,Athens

• Home page:: http://www.sequence. ...

CooVox Series IP Phone System

• No license fee
• GSM or 3G optional
• Monitor and whisper
• 10 concurrent calls
• 30 IP Phone registrations/ extensions
• Many advanced IP PBX features
• Virtual Fax, Voicemail to Email
• Call Detail Records(CDR)
• Automatic Call Distribution(ACD)
• Enhanced security policies
• H.D. Voice(G.722) supported

• No license fee
• 3G optional
• 12 concurrent calls
• 100 IP Phone registrations/ extensions
• Virtual Fax, Voicemail to Email
• Call Detail Records(CDR)
• Automatic Call Distribution(ACD)
• Enhanced security policies
• H.D. Voice(G.722) supported
• Modular design_Plug and play

• No license fee
• 3G optional
• 80 concurrent calls
• 500 IP Phone registrations/ extensions
• Firewall
• Remote backup service
• Instrusion detection
• Virtual Fax, Voicemail to Email
• Call Detail Records(CDR)
• Automatic Call Distribution(ACD)

Zycoo Argentina

VoIP Experts S.R.L. - Distribuidor Exclusivo Zycoo

exten => 100,1,Wait(2)
exten => 100,n,Record(/var/lib/asterisk/sounds/recording:gsm)
exten => 100,n,NoOp(${RECORDED_FILE})
exten => 100,n,Wait(2)
exten => 100,n,Playback(/var/lib/asterisk/sounds/recording)
exten => 100,n,NoOp(${PLAYBACKSTATUS})
exten => 100,n,Wait(1)
exten => 100,n,Hangup()

[incoming]

exten => _0[1-9].,1,GotoIfTime(9:00-18:00|mon-fri|*|*?ivr-lang,s,1)
exten => _0[1-9].,n,GotoIfTime(10:00-17:00|sat|*|*?ivr-lang,s,1)
exten => _0[1-9].,n,Playback(closed)
exten => _0[1-9].,n,Playback(closed)
exten => _0[1-9]. ...

Voice Broadcasting

• "Voice broadcasting is a mass communication technique, begun in the 1990s, that broadcasts telephone messages to hundreds or thousands of call recipients at once. This technology has both commercial and community applications. Voice broadcast users can contact targets (whether they be members, subscribers, constituents, employees, or customers) almost immediately." - Wikipedia

• Possibility to play personalized messages to every recipient which consist from library of predefined messages (pre recorder by every client) and some variable information (dates, numbers, digits, amounts with currency and so on). This method eliminates discomfort from text to voice modules (message looks completely like pre recorded by a human) and have high flexibility within predefined applications.
• List of phone-numbers to call with parameters of call in DB such as Oracle DB, Microsoft SQL, IBM DB/2, MySQL, PostgreSQL.
• Possibility to organize polls and to combine poll with statical or dynamical message. All results of such polls store within enterprise grade DB.
• Possibility to transfer to call center by DTMF command from called party.
• Possibility to opt-out by DTMF from called party.

VoiceXML and Voice Broadcasting

VoiceXML Gateway Information

See Also (Vendor Information)

Voice Broadcast Applications

• Advants IVR Technology from Stauffer Technologies, Inc. - Speech enabled voice broadcast solutions on premesis and hosted. High Quality solutions at reasonable rates!
• Call Reminders - Voice broadcast reminders
• Campus Alert Service - Notify faculty, students and community at large during a school emergency using voice broadcasting.

• ALTOTELECOM - AltoTelecom is VoIP company that provides SIP Trunking services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, rates under 1 cent per minute to USA, Canada and UK www.altotelecom.com

• Incorpus TeleNetworks Your one stop solution to all voip needs, Call termination, resellers, DIDs, Softswitches, Tollfree, residentail voip, Call center voip.

• AVOXI AVOXI Virtual Call Center Solutions - VoIP Service Provider, provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers. Contact Number 1-800-462-8694.

• CallForwarding - Be present anywhere in the world with toll free forwarding services from CallForwarding.com. Contact Number: 800-231-9802

• i7 Solutions - VOIP Wholesale and Retail i7 Solutions is looking for VoIP Resellers in markets all around the world. We provide the Dialers and Rates your customers want, and the tools for you to control all aspects of your business. i7 Solutions is a one of the very few global VoIP solution providers with direct contracts with local incumbent telecommunications operations across the globe. Our primary business includes A2Z termination services, retails & reseller solutions for VoIP services and also providing calling cards. India TATA CLI is our premium route. Please contact us @ Gtalk and Email ID info@i7solutions.in

• UtterU UtterU is leading online voice calling business service provider. UtterU provide Wholesale option for reseller to start international calling business around the world. UtterU also provide SIP trunk facility for all business to making call over the internet at very low cost.

• http://www.doorVaani.com DoorVaani.com is a Business, Residential and Personal VOIP services provider offering VOIP Call Minutes to destinations worldwide, Local Phone Numbers or DIDs in about 60 countries and Toll-free numbers in 5 countries. DoorVaani.com is also an automated web application where ordering of services, making payments and provisioning is done all in real time with no wait time or human interaction needed. Online payments are accepted in 25 currencies including Indian Rupees (INR). Offline payments or Bank Transfers accepted in INR and USD.

• Matrix ComSec - Leading SIP provider in India

VoicePulse

Company News

VoicePulse for Business

VoicePulse SIP Trunking

Rates

Account Center

• View your current Statement Balance
• Monitor real time costs for usage, Endpoints, Trunks, Gateways, Call Apps, Channels, and E911
• Make instant payments by credit card
• Add unlimited Channels or call paths to your Trunk
• View your active phone numbers
• See our inventory of numbers
• Instantly activate new numbers
• Manage E911

Supported Protocols

• Session Initiation Protocol (SIP)

Supported Codecs

• G.729a
• G.711ulaw
• G. ...

Xorcom Ltd.

• Astribank - The first channel bank designed for Asterisk.

• Up to 8/16/24/32 FXS/FXO ports per 19" 1U wall- or rack-mountable unit
• Up to 8 BRI ISDN ports
• Up to 4 E1/T1 (PRI, R2, CAS) ports
• Connects via USB 2.0, no need for additional PRI card, USB 2.0 hub may be used
• Features input and output relays for connecting and operating external devices
• Easy to install and implement

• CompletePBX with Built-in Redundancy – Blue Steel

• Various combinations of FXS, FXO, BRI and E1/T1 (PRI/R2/CAS) ports
• Supports SIP and IAX2 phones and trunks
• From 8 to 32 analog lines/extensions integrated into the unit
• Up to 4 ISDN E1/T1 (PRI, R2, CAS) ports
• Two hot swappable disk drives
• Two built-in redundant power supplies
• Support for up to four Astribank USB channel bank units internally (24 Astribanks total)
• Up to 480 concurrent calls (550 if SIP-only)
• Dual hard drive (RAID1)
• Redundant fans for cooling
• Internal backup and restore
• Front panel USB access
• Two Ethernet ports to allow separation of IP voice and data traffic for improved voice quality and increased throughput
• Supports auxiliary appliances (door locks, alarm systems). Available for models that feature I/O ports.
• FreePBX™ - Easy-to-use Web interface for Asterisk and network setup
• Advanced support and maintenance features:
  • A multi-function LCD (Liquid Crystal Display) to perform the most common functions directly on the front panel of the IP-PBX without having to attach a keyboard and monitor.
  • Rapid Tunneling™ - Provides direct support for customers via a secure connection, behind firewalls and NATs
  • Internet updates
  • Configuration export / import

• XR1000 Asterisk Appliance for SOHO