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        <title>Voximal documentation - voipproviders</title>
        <description></description>
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       <dc:date>2026-04-30T15:31:14+00:00</dc:date>
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    <item rdf:about="https://wiki.voximal.com/doku.php?id=voipproviders:appliwave&amp;rev=1475227454&amp;do=diff">
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        <dc:date>2016-09-30T09:24:14+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>appliwave</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:appliwave&amp;rev=1475227454&amp;do=diff</link>
        <description>AppliWave IntegralVoice

	*  Web site : &lt;http://www.appliwave.com/&gt;

Spécialisée dans les services hébergés innovants pour PME, Appliwave conçoit et distribue des solutions d’hébergement, de télésauvegarde, de messagerie, de connexions Internet et de téléphonie Centrex IP.</description>
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        <dc:date>2016-09-13T10:44:01+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>chan_rtmp</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:chan_rtmp&amp;rev=1473763441&amp;do=diff</link>
        <description>Asterisk Channel RTMP

	*  Web site : &lt;http://www.ulex.fr&gt;

Flash/RTMP is an add-on channel driver for Asterisk to manage any bi-directional voice or video calls from a Web browser with a Flash player. The Real Time Messaging Protocol (RTMP) is a proprietary protocol developed by Adobe Systems for streaming audio, video and data over the Internet, between a Flash® player and a server. Adobe Flash® is the industry leading web application environment, present in web browsers on 99% of the world’s …</description>
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        <dc:date>2016-07-05T22:26:43+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>dbl</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:dbl&amp;rev=1467757603&amp;do=diff</link>
        <description>DBL Technology Voip/GSM gateways

	*  Web site : &lt;http://www.dbltek.com/&gt;

DBL Technology Ltd. is a leading company in developing and manufacturing advanced voice over IP (VoIP) solutions for the worldwide market.  We are dedicated in the development of the latest VoIP technologies, and more than 10 years in developing a wide range of IP Phones and Gateways.  The offering of high quality, innovation, affordability and flexibility inside each and every single one of our products is our commitment…</description>
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        <dc:date>2016-09-07T16:29:01+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>didx</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:didx&amp;rev=1473265741&amp;do=diff</link>
        <description>DIDx Incoming calls

	*  Web site : &lt;https://www.didx.net/&gt;

DID Exchange is a VOIP-based wholesale DID Phone Number marketplace which can and is used by Mobile Operators, LECs, ClECs, IXCs or other telecommunications service providers who offer or use any kind of PHONE NUMBERS in there network.</description>
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        <dc:date>2017-02-03T22:26:32+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>iax</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:iax&amp;rev=1486160792&amp;do=diff</link>
        <description>IAX Softphone Configuration

	*  Web site : &lt;http://www.voip-info.org/wiki/view/IAX&gt;

Inter-Asterisk eXchange (IAX) is a communications protocol native to the Asterisk private branch exchange (PBX) software, and is supported by a few other softswitches, PBX systems, and softphones. It is used for transporting VoIP telephony sessions between servers and to terminal devices. The original IAX protocol is deprecated and has been superseded by a second version, commonly called IAX2. The IAX2 protocol…</description>
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        <dc:date>2017-09-13T09:39:37+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>ovh</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:ovh&amp;rev=1505295577&amp;do=diff</link>
        <description>OVH Telecom

	*  Web site : &lt;https://www.ovhtelecom.fr/&gt;

OVH est un opérateur télécom qui propose des solutions de téléphonie complètes. Commercialisées depuis 2007, plus de 100 000 lignes téléphoniques sont déjà actives. Avec les appels inclus vers les fixes et les mobiles, une vaste gamme de téléphones compris dans les offres et plus de 40 services professionnels associés, OVH se place au tout premier rang des opérateurs télécoms français.</description>
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        <dc:date>2017-11-06T15:52:42+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>ovhsms</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:ovhsms&amp;rev=1509983562&amp;do=diff</link>
        <description>OVH SMS notification

	*  Web site : &lt;https://www.ovhtelecom.fr/&gt;

OVH est un opérateur télécom qui propose des solutions de téléphonie complètes. Commercialisées depuis 2007, plus de 100 000 lignes téléphoniques sont déjà actives. Avec les appels inclus vers les fixes et les mobiles, une vaste gamme de téléphones compris dans les offres et plus de 40 services professionnels associés, OVH se place au tout premier rang des opérateurs télécoms français.</description>
    </item>
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        <dc:format>text/html</dc:format>
        <dc:date>2016-08-05T10:52:36+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>skype</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:skype&amp;rev=1470394356&amp;do=diff</link>
        <description>Skype Connect

	*  Web site : &lt;https://www.skype.com/&gt;

Receive Skype calls on your office phones and make low cost calls by integrating Skype with your SIP phone system.

By adding Skype Connect to your existing SIP-enabled PBX, your business can save on communication costs with little or no additional upgrades required.</description>
    </item>
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        <dc:format>text/html</dc:format>
        <dc:date>2015-11-15T14:22:22+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>virtualphoneline</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:virtualphoneline&amp;rev=1447597342&amp;do=diff</link>
        <description>VIRTUAL Phone Line

	*  Web site : &lt;http://www.virtualphoneline.com&gt;

Virtual Phone Line is a call forwarding service provider company from Super Technologies, that enables you to receive calls anywhere in the world through our service with reliability, On the platform such as Cellphone, Skype, Google Talk, IAX2, MSN, SIP etc, through a unique geographical number provided to you after you sign-up on Virtual Phone Line.</description>
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        <dc:date>2016-04-26T07:11:46+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>voztelecom</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:voztelecom&amp;rev=1461654706&amp;do=diff</link>
        <description>VozTelecom

	*  Web site : &lt;https://www.voztelecom.com/&gt;

VozTelecom, especialista en servicios en la nube y líder en centralita virtual y telefonía IP para empresas.

	*  Configuration  :

SIP incomming trunk


[general]
 Trunk Name: voztele-in (for example)

[Outgoing Settings]

 Trunk Name: voztele-in (for example)
 PEER Details:
   type=peer
   host=voztele.com
   username=00339xxxxxxxx
   fromuser=00339xxxxxxxx
   realm=voztele.com
   fromdomain=voztele.com
   dtmfmode=rfc2833
   canreinvit…</description>
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        <dc:date>2017-04-14T23:39:33+00:00</dc:date>
        <dc:creator>Anonymous (anonymous@undisclosed.example.com)</dc:creator>
        <title>zadarma</title>
        <link>https://wiki.voximal.com/doku.php?id=voipproviders:zadarma&amp;rev=1492213173&amp;do=diff</link>
        <description>Zadarma

	*  Web site : &lt;https://zadarma.com&gt;

Supported by a group of providers, Zadarma Project specializes in providing premium quality services. It started it&#039;s work in 2006 and continues to progress.

	*  Configuration  :

SIP trunk (with web interface)


[general settings]
 Trunk Name: Zadarma(for example)

 Outbound CallerID : (linked external number)

[Inbound Settings]

 Trunk Name: (name of the login)
  
 PEER Details:
   disallow=all
   host=sip.zadarma.com
   insecure=invite,port
   …</description>
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