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RTCP XR

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What is RTCP XR?


RTCP XR is a relatively new mechanism for compatible devices to transmit voice quality reports after the conclusion of a call. Asterisk internally has some metrics (RTPAUDIOQOS, for example) to monitor call quality, but the piece Asterisk sees is only half the picture. RTCP XR lets us measure what the end user's experience is.

Newer firmware for the Cisco SPA50x and SPA525 series phones has this capability. I am told some Polycom devices do as well, as do SNOM phones.


How does it work?


You have to set up a "collector", or system to collect the RTCP XR reports from the devices. The reports are sent using SIP PUBLISH. To my knowledge, Asterisk does not have this capability.

You can use Opensips (SIP Proxy) and a database as a collector. http://wiki.sipfoundry.org/display/sipXecs/Setting+up+an+RTCP-XR+collector+for+Polycom+Productivity+Suite gives a general how-to on the way to do this.

I found it very difficult to get Opensips installed properly until I found a YouTube video showing me how. It was much easier once I found the video and watched him run through it.

I set up my Opensips on an Amazon EC2 micro instance with Ubuntu. The only modification had to make was to add the line advertised_address="x.x.x.x" (x.x.x.x being an elastic IP assigned to the instance) to the opensips.cfg file to deal with Amazon's one-for-one NAT arrangement. Opensips receives the SIP PUBLISH notifications and passes to a simple PERL script that writes the info to a database.

See also




Asterisk cmd Ringing

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Indicate ringing tone

Description

Ringing()

Request that the channel indicate ringing tone to the user.
On an unanswered SIP channel this will send a "180 Ringing" to the endpoint.

Example

On an answered channel Ringing returns immediately and moves to the next step in the dialplan. I needed to make the caller hear ringback tone for a couple of seconds when they selected an IVR option. the solution was to use wait in combinations of ringing. See example below.

exten => s,1,Answer
exten => s,n,Ringing
exten => s,n,Wait(2)
exten => s,n,do something else

See also



Asterisk | Applications | Functions | Variables | Expressions | Asterisk FAQ

Asterisk sip progressinband

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sip.conf: progressinband


progressinband=yes

When "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio

progressinband=no

Send 180 Ringing if 183 has not yet been sent establishing audio path. If audio path is established already (with 183) then send in-band ringing (this is the way asterisk historically behaved because of buggy phones like polycom)

progressinband=never

Whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behaviour of Asterisk.

Details

Note that inband progress is not usually desired, because it requires extra Asterisk resources to run a generator to generate the inband ringing. If the endpoint is generating its own, then there is no need to tell Asterisk to do it.

Once '183 Session Progress' is sent, it is not useful to send '180 Ringing' any longer... the 183 message informs the endpoint that future progress indications for this session will be provided inband via the audio stream. Most (if not all) SIP endpoints would ignore any 180 received after a 183 anyway.

WARNING: To get inband audio on asterisk 1.6.2.x it is sometimes necessary to set prematuremedia=no. If instead prematuremedia=yes(default) is set, then inband will not work

Related ISDN example

Q: I use 2 ISDN channels with a with a fritz! card and the junghanns capi drivers. The problem appears with SIP to ISDN calls.
The SIP 180 ringing message doesn't appear because the ISDN PBX sends the "ALERT" message in-band (channel B), and not in the D channel. So Asterisk doesn't know when the ISDN channel is ringing. With my configuration Asterisk can not understand the in-band signalling for the capi channels, is it possible to use "in-band" signallisation for capi channels?

A: I guess that's "Early Media Connect", i.e., if the phone supports that (not all do), the channels get bridged just after dial completed, (SIP 183), and what you hear is the remote ring tones (from your telco), not locally generated (as if it received SIP 180 Ringing).

See also


Go back to Asterisk config sip.conf

Asterisk cmd Playback

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Synopsis

Play a sound and/or video file


Description

Playback(filename1[&filename2...][,options])

Plays the specified sound or video file(s) (you need to omit the filename extension). Sound files are stored in the /var/lib/asterisk/sounds directory by default (the directory path can be changed in asterisk.conf).

Playback is Multi-Language-compliant. It will look in a subdirectory corresponding with the current language code (as set by the SetLanguage command, or the channel's default language code. Failing that, it will play the non-language-specific edition.

Playback will play the whole sound file(s), and when complete, return control. Compare with the Background command, which plays a sound file but returns control immediately, allowing Asterisk to perform other commands on this channel while the sound file is playing.

See: Asterisk tips answer-before-playback on how to avoid the dropping of audio on the first prompt played after you answer a call.


Options

skip: Play the sound file only if the channel has already been answered. If the channel has not yet been answered, the Playback command will return immediately without playing anything.

noanswer: Play the sound file, but don't answer the channel first (if hasn't been answered already). Not all channels support playing messages while still on hook.

j: If option is specified, the application will jump to priority n+101 if present when a file specified to be played does not exist.

say: Use say.conf to interpret the string

If neither skip nor noanswer options are specified, then the Playback command will first answer the channel (if it hasn't been answered already) and then play the sound file.


Note:
The options wont work if there are spaces between the filename, the comma and the option(s).


Example

exten => 500,1,Playback(tt-weasels,skip)

Return code

On completion, ${PLAYBACKSTATUS} contains either "FAILED" or "SUCCESS".

Troubleshooting

Playback is not affected by timing, unless internal_timing is turned on in asterisk.conf.

See also

  • ControlPlayback: Play a sound file with fast forward, rewind and exit controls
  • Background: Play a sound file while processing other commands
  • WaitForSilence: Useful for example to leave a message on an answering machine/mailbox
  • Asterisk cmd WaitForNoise
  • Record: Record sound and/or video for playback
  • Playtones: Play a tone sequence

Asterisk cmd WaitForRing

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Synopsis

Wait for Ring Application.

"This application can help with phantom ringing caused by a noisy FXO line."
(quote taken from the dev mailinglist, stated in a conversation between Marius Stankevičius and Eric "ManxPower" Wieling on 03/26/2007)

Description

WaitForRing(timeout)

Waits a maximum of timeout seconds for a ring signal, which is only treated as valid until the second ring has completed.

Example

exten => 123,1,Answer()
exten => 123,n,WaitForRing(5)
exten => 123,n,SendDTMF(1234)

See also



Asterisk | Applications | Functions | Variables | Expressions | Asterisk FAQ

Asterisk cmd Progress

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Progress

Synopsis

Indicate progress

Description

Progress()

This application will request that in-band progress information be provided to the calling channel. This is also known as "early audio" and "early B3" (for example: see Dial option /b in chan_capi-cm). Note that early audio also needs to supported by your carrier in order to work for the caller.
Typically used in conjunction with Playback and its 'noanswer' option (previously 'n' option).

On an unanswered SIP channel this will send a "183 Session Progress" to the endpoint.

Starting with Asterisk 1.6.2: SIP no longer sends the 183 progress message for early media by default. Applications requiring early media should use the progress() dialplan app to generate the progress message.

Example

exten => 500,1,Progress()
exten => 500,n,Wait(1)
exten => 500,n,Playback(WeAreClosedGoAway,noanswer)
exten => 500,n,Hangup()

ISDN incoming call - inband info and announcements BEFORE ANSWER

Question: Is there a way to force Asterisk to send DSS1 PROGRESS message to PSTN with indicator: "Inband information now available", before call is established (even before ALERTING phase)? I also think that this indicator can be contained in CALL PROCEEDING message.

My idea is to play not billed welcome message on Asterisk system. Just now there is incoming SETUP, Asterisk replies with CALL PROCEEDING (without indicator I presume - but I can think only from Asterisk trace, no ISDN tester available at the moment). In ideal case there should be send PROGRESS or CALL PROCEEDING message with that indicator.

How to setup this (for PRI and junghanns.net BRI)?

Answer: Use the Progress() application in your dialplan before you Answer() the line. Use the Background() application with the 'n' flag or the Playback() application with the 'noanswer' flag to play your announcement (so the line is not Answer()'d automatically for you).

See also



Asterisk | Applications | Functions | Variables | Expressions | Asterisk FAQ

Asterisk cmd proceeding

Asterisk Paid Support

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Asterisk Consultants Germany

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Asterisk consultants: Germany


Add your entry here (Alphabetical order by country and company):
This page is growing large. Please don't post logos!!
Add your geo-position on Asterisk-users Counter.

ADDIX Internet Services GmbH, Kiel

ADDIX bietet auf Asterisk basierende Telefonanlagen, Datenbank gestuetztes Administrationstool fuer Admins und User, Echtzeitmanagement von Telefonkonferenzen, Vermittlungsarbeitsplätze, Mobile Applicationen für Smartphones. Anbindung an Junghanns BMS Callcenter Applikation, Master/Slave Systeme zur zentralen Verwaltung in groesseren VPN/MPLS Netzen mit x Systemen. Internet Services und eigene Data Center. Asterisk Hosting mit PSTN Gateway, VPN/MPLS (Filialvernetzung) und Programmierung.

ADES GmbH, Burscheid

ADES GmbH bietet von der Konfigurationsunterstuetzung bis zur kompletten ASTERISK-Anlage komplette Dienstleitungen an.
    • Home page:: http://www.ades.de
    • Telephone: PSTN +49.2174.64043
    • Email:asterisk@ades.de
    • Contact: Bent Weichert

Adimus GmbH, Bochum

Adimus bietet Beratung, Konzeption, Implementierung und Support von Asteriskbasierenden Telefonanlagen sowie VPN-Lösungen für klein- und mittelständische Unternehmen.


aixvox GmbH, Aachen

Die aixvox GmbH ist ein international tätiges Beratungs- und Dienstleistungsunternehmen. Unser Fokus ist es, Telekommunikationsinfrastruktur in Unternehmen neu aufzubauen bzw. vorhandene Strukturen zu profitablen Systemen auszubauen; Asterisk bzw. Asterisk-basierte Systeme sind oft die passende Lösung. aixvox ist auch Herausgeber des unabhängigen Kompendiums voice compass 2007, der ausführlichen Übersicht über den deutschsprachigen Voice Markt.


AMOOMA GmbH, 56566 Neuwied

AMOOMA bietet sowohl Consulting wie auch Schulungen zu den Themen Asterisk, VoIP und Trouble-Ticket-Systeme an. ...

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



News Resources

Asterisk hardware

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VOIP Service Providers Business

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This is a list of Business VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


VOIP Service providers divided by country/continent:

Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.


List of Service Providers:



www.birchills.net
More businesses in more places are making more calls using VoIP phone systems than ever. Whatever you want to say – it’s simpler and cheaper with Birchills.net – the simplest VoIP yet.

CommPeak Ltd. provides wholesale VoIP termination services to VoIP Providers, Call Centers, Calling Card Operators, Independent Software Vendors developing VoIP applications and SMB's & Enterprises. ...

Virtual PBX providers

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List of Virtual PBX Providers


Virtual BPX is a service offering functionality of a PBX without the need to install switching equipment at the customer location. Only VOIP phones need to be installed at the customer site. This makes supporting distributed workers very easy as each requires only and internet connection and a VOIP phone. A business virtual PBX phone system can reduce your monthly phone bill significantly compared to a traditional business phone system.


  • Phone2call Offers Virtual Number DIDs with Free Virtual PBX. Know why we are very known for our global leadership.
Features:
    • Receive calls from one or many local numbers
    • Forward calls to any phone number (landline or mobile), a VoIP destination or Google Talk
    • Forward calls to multiple destinations simultaneously or by predefined order
    • Record and playback voice messages to callers
    • Direct calls using extension numbers (IVR)
    • Forward calls to a voice-mail box
    • Record calls and send them to an email address
    • Send voice-mail messages to selected email addresses
    • Receive faxes and forward them to an email address
    • Host conference calls
    • Access dial tone for outbound calls (DISA)
    • Block unwanted callers
    • Notify users of selected events via email or SMS messages, with intelligent variables throughout the call flow
Advantages:
    • Extremely powerful and easy-to-use web application for managing and directing incoming phone calls.
    • Drag-and-drop objects being connected together to control exactly how the call progresses.
    • Flexibility, freedom and control over how calls to virtual numbers DID are handled.
    • Personal, and small and medium business use.
    • NO required documentation and inmediate avalaibility
    • FREE when you get your Virtual Number
    • Multilingual including English, Spanish,...

  • Smart Voice Network offering service to Carriers and Call Centers with unique ability to cap rates and manage sub accounts. VoIP Carrier with US & InternationalCanada$.01UK$.01USA Termination and Origination offers termination & origination at $0.01 per minute and DID's for $1.00__ per month all with unlimited channels / concurrent calls. Competitive High Quality NPA-NXX rate deck, International Carrier rates, and DID channel pricing. For calling cards, mobile dialer, Efax solution, Bulk SMS and white label for the bes web conference we will host it for you and payment solution Call us at 760-517-6724.
  • Business Voip
  • Hosted Voip
  • Calling Cards
  • SMS Voip
  • Internet Fax
  • Mobile Dialer

VOIP Service Providers B2B

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Here is a list of VOIP Service Providers focusing on Business-To-Business services. This includes VoIP origination and VoIP termination, plans aimed at call centers, IVR providers and generic Asterisk users. See also:

Services which require the use of locked ATA devices should not be listed on this page. Nor should services which do not permit simultaneous calls — most services here support at least 4 simultaneous incoming calls. Please list only services which support Asterisk connections, via SIP or IAX2, to the PSTN.

VOIP PBX and Servers

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Please list information about VoIP PBX and Servers on this page. Please keep VoIP PBX and server provider information in alphabetical order, and below any other relevant information.

Numeric

  • 1comms.co.uk:Asterisk-based converged telephone system for UK Businesses
  • 1GATE VoIP PBX by Wangate: Cheap VoIP PBX with hardware DSP. Optional internal gateways for Analog/ISDN/PRI/GSM. VoIP resellers welcome.
  • 2N NETSTAR PBX, virtual PBX: VoIP PBX system
  • 2N Omega IP PBX: VoIP PBX system
  • 2N VoiceBlue Enterprise: Simple VoIP SIP PBX
  • 3CX: Windows IP PBX / VOIP Phone system
  • 4PSA VoipNow: Hosted PBX software for service providers and enterprises, accelerating SaaS deployment. It runs on Linux environments (RHEL, SuSE Linux, CentOS, Fedora) on x86 and Power PC architecture based servers.
  • 8ix Zenith: 8ix Zenith spells an Asterisk derived IP Telephony application with the most advanced calling and communication features.


A

  • ActivePBX™ | Turn-Key Business Phone System $149/mo.
  • Aculab Cloud is a framework that enables developers to add telephony to their applications
  • Aksys Networks offers the KONNECT Office Phone System which unlike phone systems requires no PBX, server or Hosted service. The phones create an adhoc network that shares phone services in an office or seamlessly across multiple sites. ...

VOIP Service Providers Business North America

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This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.

Please describe services in neutral language and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!

Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.

If you like this page, please link to it, so Google and other search engines will consider it more important.

Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page.

Providers in other countries/continents can be found here:


Canada

VoIP Providers USA

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This is a list of VoIP providers in the USA. These companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP. Please add VoIP Providers USA to the list below.

  • 1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.
  • 45meeting Simple online conference solution for small business.
  • Affinity SIP is a Business VOIP service provider. Great rates for international VOIP calling U.S. DIDs $1.00, Inbound/Outbound $0.009/min, E911 and CNAM free! Wholesale SIP Trunks. Service works with all IP PBX including Asterisk, FreeSWITCH, SIP, IAX2, Elastix, trixbox®, FreePBX, and more. Redundant gateways for 100% up-time. Try SIP Affinity now risk free! Instant activation with free test credit.
  • ActivePBX™ | Turn-Key Business Phone System $149/mo.
  • 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 capability on 100% of our DIDs, CNAM, e911, and local number portability. Wholesale SIP
  • Big Bang Systems provides on premise and hosted pbx and ivr solutions for company with 50+ employees.
  • Bluestone Communications provides enterprise class hosted PBX services with QOS measurement from our servers to your site for a fundamentally better service. Extensive array of facilities and apps, just $15 per phone per month.
  • Cheap VOIP Inc Wholesale to the public, great rates and good quality.
  • DIDforSale Buy unmetered Inbound SIP DIDs for just $7.50 all over us. All DID comes with 20 channels.
  • DLS Hosted PBX offers the most call center and integration features
  • CheapVOIP4U SIP termination starting at $0.0027 per minute without fees. Pay as you go service.
  • CircuitID provides high-quality Wholesale VoIP Termination, Origination, Toll-Free, and Microsoft Lync SIP Trunking services, at a low affordable cost. Experience carrier class quality, and savings benefits.
  • Flowroute LLC Wholesale VoIP, A-Z SIP Termination, Cheap DIDs, Free CNAM Storage, E911, T. ...

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.

January 2013


December 2012

VOIP Service Providers Business Europe

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This is a list of VOIP Service Providers who offer full service products primarily aimed at the small to medium sized business telephone market. Such companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP (as an alternative to Centrex service). See also:


Service providers operating in more then one country are listed under each country. "Operating in a country" means a provider that has billing and support staff located in the country, and offers service subject to the regulations and consumer protection laws of that country.

Please describe services in neutral labanguage and normal fonts. Don't bother listing prices--unless you really plan to return and edit them as things change. If you want to add your company, please read the Posting Guidelines for Promoting Products and Services. When you add your entry to this page, please make sure your entry is in alphabetical order in relationship to other vendors listed in the same section. Failure to follow these guidelines will result in deletion!


Please include relevant information like SIP or IAX handoff, how outgoing CLID is set, whether a given account may originate multiple calls at once, etc.

Users: Please feel free to REMOVE any listing that does not meet the stated goals of this page. ...

WombatDialer

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WombatDialer is a platform to provide mass outbound calling on the Asterisk PBX.

This can be used to implement many different services. By offering you a set of ready-to-use components and a monitoring GUI, you can create complex solution in minutes.

Typical usage scenarios include:

  • Queue call-backs: allow callers to leave their numbers instead of waiting on a queue, and then being calls them back
  • Web-based contact-us forms: allow customers to be called back by using a web form
  • Automated surveys / Reverse IVRs: call a list of numbers and gather information that is entered as DTMF
  • Number verification services: automatically check the validity of phone numbers
  • Telecasting of pre-recorded messages: send pre-recorded messages to thousands of recipents quickly and efficiently
  • Automated appointment reminders: remind and take appointments, with our without human interventions
  • Automated subscription expiration: automatically call expiring subscriptions, and have clients renew their subscription
  • Automated quality surveys: run quality/performance interviews and feed them back into the QueueMetrics QA module
  • Generic progressive dialing: keep your outbound agents busy by having a dialer compose numbers, handle retries and pre-qualify leads

WombatDialer can work on pre-defined call lists or can dynamically create them over an API (e.g., dial number X after 10:30 AM). It shares the load on one or more PBX servers and has a flexible rescheduling logic to handle missed calls. It is built to be used with your existing Asterisk PBX and does not require separate servers or a separate set of lines. It can call over VoIP or through the public telephone network. It can also act as a power-dialer for a set of agents logged in in Asterisk.

WombatDialer is built to integrate with your business processes, and can receive calls to be made over HTTP and/or notify an external system in real-time of calls made and results gathered.

WombatDialer works natively with the QueueMetrics Call-Center Monitoring Suite in order to produce state-of-the-art campaign analyses and insight.

WombatDialer is currently available as a commecial product, but it is free to use for smaller systems, telephony hackers and Asterisk enthusiasts.
The current version is 0.6.5 as of January 2013.

See also:


Dispayed {HITCOUNTER} times.

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