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DID Service Providers

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A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

Cheapest DID Providers

see Cheapest ATAs and Service
see domestic USA DIDs for carriers

Free Service Providers Only (Free DIDs)


Algeria

  • Algerian DID numbers Currently only national Algerian numbers.. Also 68 other countries available. Free forwarding to Voip/SIP, IAX,H323 Google Talk, etc many providers preconfigured. Forwarding to PSTN (landlines and mobiles) from 2cent/min, including mobiles. Mobiles Europe 6cent/min.

Argentina

  • Phone2call Virtual Numbers DID are available in 60+ countries including Argentina. Free forwarding to Voip/SIP (some preconfigured providers), IAX, H323, Google Talk, Virtual PBX (yes, free Hosted PBX). Also, very cheap call forwarding to PSTN (regular landlines and mobiles). No contracts. Pay per month, Instantaneous Activation, Many payment methods. Try not only our Virtual Numbers DID but also our free Hosted PBX which has many features (useful for example for Small and Medium Business). Know why we are known for our VOIP leadership.
  • Argentina Virtual Numbers $5.95 Per month / Free Setup| Virtual Numbers in Argentina with Free Forwarding Options(FlyNumber)
  • Argentina DID Numbers€ 4.99/month| DomesticNumbers offers Argentina virtual phone numbers from 19 different cities in Argentina, including Buenos Aires. Website also in Spanish.

Australia


Austria


voip-info.org

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


This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.

Update: We have added theFacebook Like and Google +1 button to the top right corner of all pages on Voip-Info.org. Please help recommend the wiki by clicking on them. We also now have Google+ page here and a Facebook page here. Visit them and add us. Thanks!


NEWS

Asterisk call queues

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Queues consist of
  • Incoming calls being placed in the queue
  • Members that answer the queue (extensions or users that login as agents)
  • A strategy for how to handle the queue and divide calls between members
  • Music played while waiting in the queue
  • Announcements for members and callers

Queues are defined in queues.conf or in dynamic realtime. The latter allow storing queue configuration in a database so that changes are immediately available for new callers without the need for an explicit reload.

  • Agents are the people (or person) that answer call(s) that have been placed into a specific Queue. An agent logs in indicating that s/he is now ready to take calls. Asterisk transfers an inbound call to a queue, which is then in turn transfered to an available agent.
  • Members are those channels that are active answering the Queue. It can be agents or normal channels, like "sip/snom23"

New in Asterisk v1.2

A queue is now considered empty not only if there are no members but also if none of the members are available (e.g. agents not logged on). To restore the original behavior, use "leavewhenempty=strict" or "joinwhenempty=strict" instead of "=yes" for those options.

It is now possible to use multi-digit extensions in the exit context for a queue (although you should not have overlapping extensions, as there is no digit timeout). This means that the EXITWITHKEY event in queue_log can now contain a key field with more than a single character in it.

Members

Members can be direct channels, i.e. phones connected to Asterisk. You can also define members as individuals that login from any connection to receive calls.
Agents are defined in agents.conf. Agents login from other phones on special extensions that use the agentlogin application.

Strategies

Calls are distributed among the members handling a queue with one of several strategies, defined in queues.conf
  • ringall: ring all available channels until one answers (default)
  • roundrobin: take turns ringing each available interface (deprecated in 1.4, use rrmemory)
  • leastrecent: ring interface which was least recently called by this queue
  • fewestcalls: ring the one with fewest completed calls from this queue
  • random: ring random interface
  • rrmemory: round robin with memory, remember where we left off last ring pass



Menu for the user

You can define a menu for the user, while waiting. For this menu, you can only use one-digit extensions (Please, read this : a couple lines above, it is mentioned that, starting from 1.2, multi-digit exit are allowed. Can someone correct this ?). Define the context for the menu in the configuration for the queue to enable this option.


Penalties

Queue members can be defined as having a penalty - e.g.
member => SIP/200,1
member => SIP/201,2
member => SIP/202,3
member => SIP/203,2
If the strategy is defined as 'ringall', then only those available members with the lowest priorities will ring. In the example above, if 200 is not busy, then only 200 will ring. If 200 is busy, then only 201 and 203 will ring. If 200, 201 and 203 are busy, then 202 will ring.

Note: If extension 200 does not pick up it will not automatically go to extension 201. It will keep ringing 200 until they pick up. It will only go to the next extension if the current extension is either busy or unavailable.


Cascading Queues

You can set up a series of queues that cascade to each other. ...

LUXMS - V2CHAT

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LUXMS INC. is an American technology company in Wilmington, Delaware. Our R&D Center is located in St. Petersburg, Russia.

Our team offers V2Chat: white label and value adding web VoIP service for B2C companies, carries, and VoIP providers.

The solution consists of Flash-to-SIP signaling and media gateway and customizable widget – V2Chat web call button (Flash+Ajax).

Buttons are placed on client websites. Calls are made by website visitors by a click. Website visitors don't have to install additional software to make calls. Calls are made onsite - right from the browser.

V2Chat - an optimum solution to improve customer care and increase customer satisfaction rates.

V2Chat enables:

  • voice and voice/video calls;
  • call forwarding to PCs, landline and mobile phones;
  • deeply customizable look and feel;
  • adjustable working hours.

Additional features can be included upon request:

  • branding;
  • call encryption;
  • Design Studio for advanced customization;
  • access to V2Chat API.

Key Advantages of V2Chat:

  • Brand Experience;
  • Cost efficiency;
  • Flexibility.

V2Chat service is rendered directly to clients. Luxms Inc. is also open for business cooperation with carries and other VoIP operators.


Free two-week trial can be ordered here.


VOIP Service Providers Residential

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chan_mobile

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chan_mobile (used to be chan_cellphone) — Use Bluetooth cell / mobile phones as FXO devices


Asterisk Channel Driver to allow Bluetooth Cell/Mobile Phones to be used as FXO devices and Bluetooth Headsets as FXS devices
The official Homepage is http://www.chan-mobile.org (no longer available).

Features :- (Oct 2007)
  • Multiple cell phones can be connected (subject to some limitations - see Notes).
  • Multiple bluetooth adapters can be supported.
  • Asterisk automatically connects to each cell phone when it comes in range.
  • Command to discover bluetooth devices. Useful for configuration. Requires an unused bluetooth adapter.
  • Inbound calls to the cell phones are handled by Asterisk, just like inbound calls on a Zap channel.
  • Caller ID passed through on inbound calls.
  • Dial outbound on a cell phone using Dial(CELL/device/nnnnnnn) in the dialplan.
  • Use a Bluetooth Headset as extension using Dial(CELL/device) in the dialplan.
  • Application CellStatus can be used in the dialplan to see if a cell phone is connected.
  • Application MobileSMS to send SMS via a connected mobile phone
  • Supports devicestate for dialplan hinting.

chan_mobile was written by David Bowerman and is officially supported only for the Asterisk development trunk and is available as an add-on from http://svnview.digium.com/svn/asterisk/trunk/addons/chan_mobile.c?view=markup. Current documentation can be obtained from http://svnview.digium.com/svn/asterisk-addons/branches/1.6.2/doc/chan_mobile.txt?view=markup
Unofficial and unsupported Asterisk 1.2 backport can be obtained from http://www.sigsegv.cx/sip-9.html.

Note:The above seems a bit out of date (24-11-2011). Branch 1.6 includes chan_mobile in Asterisk Addons, and from 1.8 onwards it is included in the main Asterisk source download (in the add-ons config menu).

A good installation guide for trixbox can be found under http://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html.

Note: FC6 bluez yum packages with latest updates will not work, see comment below
Note: Some mobiles (Motorola V3 and K1) report themselves as a valid headset, but they do not work if you configure them as a headset in mobile.conf (not fully tested)
Note: Not all mobiles with bluetooth profiles have all the features necessary for this channel to work. For example - Nokia E65 is not usable.
Note: Each mobile "eats" one bluetooth adapter. Multiple mobiles cannot connect to the same adapter so if you want to connect multiple mobiles prepare to buy dongles by the basket.

SMS

In chan_mobile.c, you'll see apps MobileSendSMS(device,dest,message), which allows you to send an SMS message via the dialplan, thru the bluetooth attached phone.

To get an SMS, you have to have a cellphone bluetooth attached, and capable of passing sms messages. When it reports to Asterisk via the bluetooth connection, that an SMS message was recieved, Asterisk will try to run the "sms" extension, with the channel variables SMSSRC and SMSTXT channel variables set to the appropriate values. In the dialplans you can turn this into an email, an announcement, a text-to-speech (via festival or Cepstral or whatever), or whatever your needs or imagination can supply.

I've asked around a while back, and the only phone capable of such sms capabilities was one running the Symbian OS. ...

IPsmarx Technology, Inc.

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Learn more about IPsmarx VoIP Solutions!
We deliver comprehensive and turnkey, customer management and billing solutions for VoIP Service Providers, Carriers, VoIP Calling Card Operators, and ISPs. Our Solutions enable Internet Telephony Service Providers (ITSPs) and entrepreneurs to deploy and run different VoIP business models with all the necessary tools.

Our portfolio includes:

Prepaid and Postpaid Pinless Calling Card Solutions
Softswitch IP-IP Billing Solution
CallShop and Hosted CallShop Solution
Wholesale Carrier Solution
A-Z Wholesale Carrier Services
SIP Enabled Cell Phone Support
Multi Tenant IP PBX
Class 5 softswitch

We take pride in providing highly efficient, modular, scalable, and reliable solutions at very competitive costs that result in rapid return of investments.

For more information visit our website at:http://ipsmarx.com/ or contact us at: sales@ipsmarx.com

Asterisk consultants Asia

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This is a page about Asterisk consultants in Asia. Please keep this list alphabetical.

Asia

BANGLADESH

E-NET COMMUNICATION

We provide the following consultancy:

    • Asterisk SS7/E1/T1 Solution
    • Asterisk GUI
    • Asterisk AGI programing
    • Asterisk Based Call-Center
    • Asterisk Calling card/callback
    • Asterisk IPPBX
    • OpenSIPS/kamailio/Yate/GnuGK Consultancy
    • Asteisk Billing
    • Asterisk at large environment

Please contact:
    • Contact: Mohammad Emran
    • Phone: +880-1713033045, +880-1768200000
    • Address: HB Tower(3rd Floor),House No# 1/A,Road # 23, Gulshan-1,Dhaka-1212,Bangladesh.
    • Email:mailto:monemran@gmail.com


CHINA/SINGAPORE/MALAYSIA/HONG KONG/TAWAN

BangJian telcom Ltd.

BangJian telcom Ltd located in ShenZhen, China. BangJian Telcom provides PBX, CRM, Call center, SS7/PRI solutions and asterisk cards. Lists:

CHINA


asterCC technology Co., Ltd.

asterCC is a software development company, our main product is asterisk based IP PBX & Call Center system.
We have been active in software development for computer telephony and Internet, for over 5 years.


    • asterCRM open source call center & crm
    • asterBilling hosted call shop & realtime billing solution.
    • asterCC Commercial hosted PBX & Call Center solution, provides free ip pbx features and 5 free agents license.

WShuttle Infotech Ltd. ...


VOIP Billing

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Hosted Billing Services (in Alphabetical Order)

  • Adore All-in-One SIP Server and Client v2.2.1 - new released with Class5 features
  • Aradial AAA for Billing Solutions
  • BillCall - Telecom Resource Management for wholesale Voip Carriers Panamax’s Telecom Billing Solution BillCall provides solutions for End-User billing, Carrier Access Billing (CABS), CDR Mediation, Rating & Routing.
  • CCM Billing Affordable CDR billing solution for Cisco CallManager telephony systems.
  • Cybercallshop ultimate callshop server Incredible advanced online callshop server 100% standalone able to handle many shop, simply the best software to get customer loyalty because it's many more than simple online booth billing.
  • Cyneric Fully Integrated Billing Solution Platform. Compatible with: Cisco, Radius, Mera, SIP, SER, Asterisk, Quintum, SNOM, Audiocode.
  • DORETEL Communications, Inc. Hosted Calling Card Solution, Hosted Wholesale VoIP Billing Platform, Hosted SoftSwitch Solution, Hosted Call Shop
  • DTH Free Call Rating Service was designed for companies who need to rate their customer's calls but can't yet justify implementing a full VoIP Billing and Customer Management system.
  • dtlvoip is hosted VoIP billing and switching service provided by DataTechLabs since 2002.
  • Call Shop Billing We are leading providers offering call shop billing solutions. We offer our services to many businesses and residential clients who wish to make international calls. Visit us now to test our services.
  • Dynasoft TeleFactura Outsourced - Full ASP outsourced billing services for the Telecoms industry with suites of Web applications made available to your end-users.
  • EasyITSP Open Source ITSP and Hosted PBX software for Asterisk
  • EZ Calling Card Hosted VoIP calling card platform with premium A-Z VoIP termination.
  • Haagenti Group Inc has operations in Berlin providing A-Z wholesale termination, hosted VoIP applications, call forwarding, LCR and intelligent call routing as well as VoIP and GSM carrier (TAP3) billing
  • Inovaware Corporation - VOIP-Pro: fully hosted VOIP billing, rating, provisioning, help desk and customer care solutions. Flexible, scalable and robust architecture for billing voip carriers of all sizes.

VOIP GSM Gateways

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What's a VoIP GSM Gateway?

A VoIP GSM Gateway enables direct routing between IP, digital, analog and GSM networks. With these devices (fixed cellular terminals) companies can significantly reduce the money they spend on telephony, gp-especially the money they spend on calls from IP to GSM. The core idea behind cost saving with VoIP GSM Gateways is Least Cost Routing (LCR).

Through least cost routing the gateways select the most cost-effective telephone connection. They check the number which is dialed as well as rate information which is stored in an internal routing table. Because several SIM cards and GSM modules are integrated within the VOIP GSM Gateway it is able to make relatively cheaper GSM to GSM calls instead of expensive IP to GSM calls.

Who offers VoIP GSM Gateways?


Web Hosting

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Please add information about web hosting and web hosting providers and companies to this page.

Web Hosting Providers


Please keep this list in alphabetical order

H.323

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H.323 is an ITU VOIP protocol. It was created at about the same time as SIP, but was more widely adopted and deployed earlier. Today, most of the world's VoIP traffic is carried over H.323 networks, with billions of minutes of traffic being carried every month.

H.323's strengths lie in its ability to serve in a variey of roles, including multimedia communication (voice, video, and data conferencing), as well as applications where interworking with the PSTN is vital. H.323 was designed from the outset with multimedia communications over IP networks in mind, making it the perfect solution for real-time multimedia communication over packet-based networks.

  • 2N H.323/SIP Gateway
  • ITU H.323 Page
  • Packetizer's H.323 Information Site
  • OpenH323 channel driver: asterisk-oh323
  • Abilis Abilis the all-in-one VoIP gateway with ISDN backup
  • Asterisk H323 channels
  • ATcom:H.323 to ISDN Gateway
  • Ekiga H323 Ekiga, formerly Gnome-meeting, supports H323
  • Open H.323: Open Source implementation
  • OpenH323 Gatekeeper: The GNU Open Source H.323 Gatekeeper
  • ISDN2H323: H323 to ISDN Gateway (discontinued)
  • IsdnGw: H.323 to ISDN Gateway
  • ooh323c: An Open Source C implementation of H.323 stack
  • Uniqall Gridborg HMP Proprietary Host Media processing server with H.323 and SIP frontends, and simple ASCII control protocol. It works in both Linux & Windows environments. Its client-server architecture enables you to use any programming or scripting language. It can handle 240 ports on dual processor servers.
  • Yate it's free software (open source) that use OpenH323. The H.323 channel in Yate it's considered to be the best free implementation based on OpenH323. Yate also works as a SIP-H323 signalling proxy, for companies who have internal SIP networks and H.323 carriers.


H323 Variables

External H.323 links

SIP method prack

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PRACK is defined in RFC 3262: Reliability of Provisional Responses in the Session Initiation Protocol (SIP)


The PRACK request plays the same role as ACK, but for provisional responses. There is an important difference, however. PRACK is a normal SIP message, like BYE. As such, its own reliability is ensured hop-by-hop through each stateful proxy. Also like BYE, but unlike ACK, PRACK has its own response. If this were not the case, the PRACK message could not traverse proxy servers compliant to RFC 2543.





Numerous implementation problems seen in the field
A SIP UA indicates support for this standard by including a "Supported: 100rel" or "Require: 100rel" as a SIP header. Several major SIP stacks — including the one in IOS and on PolyCom SoundPoint IP 500 phones — have shown problems with it, at least in previous versions. The SIP headers claim to support it or require it, but when you send them a non-100 1xx message, they don't PRACK it.




Old News

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This page lists all the old VoIP news stories from the home page.

November 2012

Call Center CRM by Voicent

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Voicent's Call Center CRM is a powerful software tool for managing customer interactions in call centers or sales organizations.

Use it to:


  • Automatically track all customer interactions!
  • Inbound calls and SMS, outbound voice, text messaging or email campaigns--all activities are automatically saved to individual customer record.
  • Automatically save and manage Customer Opt-Out and Contact Preferences
  • Records customer opt-out selections for your auto dialing or predictive dialing campaigns; auto saves customer choices for preferred level of contact or mode of contact (phone, text, email, etc.) in IVR application when they call in.
  • Automatically display customer information to agents
  • Fully integrated with agent desktop software, Call Center CRM increases organizational intelligence and efficiency by automatically displaying customer information, such as contact and buying history, when a call is connected to an agent.
  • Intelligently, efficiently recognize callers and serve them accordingly
  • Automatically direct callers to the appropriate department, agent, Interactive Voice Recognition menu, recorded-message language, custom welcome message or however you'd like to personalize their experience when calling you.
  • Design and manage sales and marketing campaigns
  • Search customers and automatically create new outbound campaigns based on location, buying preferences, sales activity, purchasing history or other differentiation.

Call Center CRM is great for:


  • Helping organizations identify and better understand customer needs by keeping track of customer interactions and displaying them right at the point of future contact.
  • Providing a professional and credibility-enhancing contact experience for prospects and customers.
  • Designing and managing various marketing and sales campaigns by saving call status and customer choices (such as opt-out) automatically.
  • Serving customers better by automatically sharing customer contact information among different agents of an organization, increasing organizational intelligence and efficiency.

Key Features Of This Call Center CRM Software


  • Works seamlessly with Voicent's Agent Dialer predictive dialer, BroadcastByPhone auto-dialer, and IVR Studio Interactive Voice Response app designer.
  • Automatically keeps track of customer interactions by phone, text message, and email.
  • Automatically remembers opt-out selection and prevents calls to the opt-out phone numbers in the future.
  • Manages customer profile and contact history with customizable customer profile fields and display options.
  • Automatically displays customer information for both inbound and outbound calls when calls are connected to agents.
  • Lets you easily search customer lists and automatically create auto dialing and predictive dialing campaigns.
  • Easy to use - download and setup Call Center Manager in less than 5 minutes.

What do I need to use call center CRM?


  • A desktop or laptop computer with Windows 2000/2003/XP/2008/Vista/Windows 7/8 operating system.
  • Voicent Agent Dialer predictive dialing software, and/or BroadcastByPhone auto-dialer software, and/or IVR Studio, and/or Flex PBX software-based IP PBX system.

How to use Call Center CRM?


1. Simply install the Call Center CRM and enable it from the Voicent Gateway. Once enabled, it will run in the background to keep track of all interactions.

2. From the Call Center CRM program window, you can:

3. Create, edit, and search customer information.

4. Create outbound campaigns for BroadcastByPhone, AgentDialer, BroadcastBySMS, or BroadcastByEmail.

5. Import customer profile information.

It's absolutely free to download and try Voicent's Call Center software. You don't need a credit card or even to supply an email address to try it. ...


Voicent Communications

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Image

Voicent Communications, Inc provides affordable and easy-to-use telephony and email communication solutions. Voicent is dedicated to developing affordable, easy-to-use communication software that helps people, businesses and communities share information, exchange ideas and build strong relationships. Voicent's innovation always aims to simplify the complex --to make technology simply work wonders. Voicent's products provide voice communication, email communication and text-message communication with unparalleled levels of automation and point-and-click ease. Through our Developers Network, we also offer our advances in software development to other software professionals, enabling them to integrate sophisticated communication capabilities into their own tools and applications.

  • Voicent has implemented many of the telephony hardware based features in software.
  • This technology enables feature rich telephone systems to be totally based on off-the-shelf voice modems. For example, call progress detection is usually a hardware feature that is not available in voice modems.
  • With Voicent software, call progress detection and human vs. answering machine detection are all available with voice modem based systems.
  • This technological implementation drasticaly reduces the cost of these systems.
  • Voicent software works with regular telephone, skype, and SIP based VOIP services. Release 8 also support SMS/text message.

Voicent provides the following products:

1. Call Center Software - Powerful software tool to automatically monitor and manage regulatory compliance and call center training and efficiency.
2. BroadcastByPhone Autodialer - Automated dialing software can reach people by phone or by voice mail with personalized messages.
3. Predictive Dialer - Automatically hands you only the calls answered by a live person; skips or leaves messages on machines.
4. IVR Studio - Quickly build interactive phone applications that are fully integrated with your business.
5. PBX FlexPBX - Affordable, feature-rich, telephone system with auto attendant and music onhold.
6. Broadcast By SMS - Delivers custom text messages to large or small groups and collects responses.
7. Broadcast By Email - Delivers custom email in text or HTML formats. Track open, manage bounce and unsubscribe emails.
8. Auto Reminder - Integrated appointment scheduler and automatic phone, text, and email reminder with confirmation.
9. Call Center CRM - Powerful software tool for managing customer interactions in call centers or sales organizations. ...

VOIP Resellers

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VoIP Resellers


This is a list of VoIP resellers.

1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.
  • Custom portal for your customers
  • Actual US CLEC
  • Set your own pricing for your customers

8774e4voip.com 8774e4voip.com - Contact Us Toll Free @ 877.434.8647

Air21group.co.uk - UK BASED/ 247 SUPPORT - Call us on 0121 314 1114
  • Add to your telecoms business or start your own business reselling business telecoms today with the Air 21 Group
  • Whitelabel reseller program
  • High Commissions
  • Wholesale VoIP origination
  • Termination SIP Trunking
Click to enquire now

Alcazar Networks Inc. is offering Wholesale Origination / Termination / Free Toll Free Termination - Get paid for your toll free traffic! We provide high quality, dependable access to over 3100 rate centers and instant access to over 1,200,000 DIDs including T38 and local number portability. Wholesale SIP

Call Shop - Become a Voip Reseller. We are leading provider of voip services. Start your own voip reseller business and provide voip services to your customers.

DIDForSale provides high-quality
  • Wholesale VoIP Origination,
  • Toll-Free, and
  • Termination SIP Trunking services.
  • Offers 8000+ rates center at the best price cost. Best products for Calling Card, Call Conference companies.

Phone2call | Phone2call is characterized by its leadership in the global telecom industry. Among other services, it has a reseller program which is a great opportunity for enterpreneurs/internet related companies to grow at joining us to use our full of features powerful platform and selling their own branded telecom services in their own countries/regions.
Features:
  • Own branded reseller program.
  • Very low prices per minute/SMS for resellers.
  • Create your own tariffs/rates for your users.
  • FREE use of our platform. Reliable platform.
  • FREE and active support and training by our team.
  • Offer online or material calling cards for your users.
  • Use DIDs.
  • Offer subscriptions (flat rates,...) for your users.
  • Call shop.
  • Payment gateways through your users may pay such as Paypal, Webmoney, Paypal Pro, Google Checkout, 2Checkout, and more. ...

New Software Releases

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This page is to inform on various VoIP related software releases.

Your contributions are welcome, but please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.


December 2012


November 2012

NFAS

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NFAS (Non-Facilities Associated Signaling) is an ISDN feature for sharing one ISDN D channel accross multiple ISDN PRI lines.

Example, if you used NFAS to share one D channel accross 10 PRI lines, you would gain 9 extra B channels over a configuration that used one D channel per PRI line.

NFAS also supports a backup D channel if should the first one fail.

NFAS is configured in zapata.conf



Trunk groups are used for NFAS or GR-303 connections.

Group: Defines a trunk group.
       trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]

       trunkgroup  is the numerical trunk group to create
       dchannel    is the zap channel which will have the
                   d-channel for the trunk.
       backup1     is an optional list of backup d-channels.

trunkgroup => 1,24,48
trunkgroup => 2,72,96

Spanmap: Associates a span with a trunk group
       spanmap => <zapspan>,<trunkgroup>[,<logicalspan>] 
                            - Note the logicalspan numbering must start at 0 (according to Digium)

       zapspan     is the zap span number to associate
       trunkgroup  is the trunkgroup (specified above) for the mapping
       logicalspan is the logical span number within the trunk group to use.
                   if unspecified, no logical span number is used.

spanmap => 1,1,0
spanmap => 2,1,1
spanmap => 3,2,2
spanmap => 4,2,3



The spans are registered in asterisk by the zapspan that the primary d channel resides.

Archived example (confusion due to incorrect logicalspan numbering):

spanmap => 1,1,1
spanmap => 2,1,2
spanmap => 3,2,3
spanmap => 4,2,4

In the above sample, trungroup 1 would register as span 1 and trunkgroup 2 would register as span 3.

The bolded statement above is important to understand. Let's assume we're bringing in 4 T1 channels into a 4 port Digium card. You can place the D channel on any of the T1's, but the spanmap will be different based on the one you pick.

For a D-Channel on T1 1, channel 24, you can use the example spanmap. However in our configuration we have the D-channel on T1 4, channel 24 (aka Zap-96). This requires us to change the spanmap to the following (we're starting at logical span 0 since that's what GBLX has listed):

spanmap => 1,1,3
spanmap => 2,1,1
spanmap => 3,2,2
spanmap => 4,2,0

When we had it configured 0,1,2,3 any calls that came into spans 1 or 2 worked fine. But spans 0 and 3 did not pass any audio. Switching the logical span numbers fixed it. This is because zapspan #1 is suppose to be the one that the D-Channel is on. In our case that was actually logical span 3. Be sure to watch out for this, we were confused and took up a few hours of a GBLX tech's time to get it fixed.

See Also


ISDN
PRI
Asterisk PRI

Secure VOIP Service Providers

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Marketing is NOT ALLOWED on this page. Please describe services in neutral language. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. Do not use UPPER CASE, do not use bold face, do not use italics, or exclamation points. Do not use the word "best". Please indicate whether you are a facilities-based provider or a reseller. Use neutral language and play fair. Please place your entry in alphabetical order with respect to other vendors in the same section. And if you make your own category, it will be deleted.


North America

  • Kryptotel Kryptotel protects privacy from any form of interception. products: encrypted phone, encrypted computer, encrypted vpn, encrypted email. (works all over the world)
  • CI Gear CI Gear Business VoIP services include Hosted Phone Systems and Business VoIP Lines delivered securely over the Internet to your VoIP Phone, VoIP Adapter, VoIP Phone System or VoIP Gateway.
  • SECURVOX SECURVOX provides secure telephone service for private and business use.
  • VERSAFON Secure VOIP phone calls (ZRTP) and conferencing, wide range of hardware supported . Custom solutions available.


Middle East & Asia Pacific

  • VOIP Service Provider in India - Spectranet leading Internet Service Provider Company in India offering VOIP Services in India, International Calling Cards, Prepaid VOIP Services, Business VOIP Services and all other affordable VOIP solutions in India

  • VPN4VOIP.COM Many ISP in Egypt, Iran, Pakistan and Bangladesh etc. filter out VOIP traffic and block calls from VOIP gateways, VPN4VOIP.COM is a professional VPN service dedicates to solve this headache. It's an industry leading SSL based VPN for VOIP Service Provider widely used by Voice Over IP peers and carriers all over the world to unblock wholesale VOIP traffic since Year 2006. By utilizing highest secure VPN technology, it allows VOIP provider to bypass ISP's ban on H.323 or SIP VOIP calls from Cisco AS5300 or Quintum Tenor AX gateway, mapping static public IP to AudioCodes or Mediatrix gateways for convenient setup and establishing QoS guaranteed VLAN based VPN connection with dedicated Tier-1 bandwidth for high voice quality VOIP origination and termination minutes exchange business.
  • Kryptotel Kryptotel protects privacy from any form of interception. products: encrypted phone, encrypted computer, encrypted vpn, encrypted email. (works all over the world)

South & Central America

  • Kryptotel Kryptotel protects privacy from any form of interception. products: encrypted phone, encrypted computer, encrypted vpn, encrypted email. ...
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