1TM Guatemala

Expertos en Linux y telefonia IP Asterisk.

Marcas:
VIOPBX, Sangoma, Digium, Polycom, Cisco, Linksys, Astra, Bria, Eyebeam, y mas.

Your rate:
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General office

1TM
12 calle 12-76 zona 10 Colonia Oakland
Guatemala
Guatemala ,01010
Guatemala

Phone: +50223296280
Site: http://www.i-t-m.com
Contact Person

Manuel Andrade / Gerente Comercial
Phone: +50223296280 EXT.400
  • Cloud-based Unified Communications Service Provider
  • did numbers provider
  • FREE Callshop Software
  • Hardware
  • Hosted PBX Service Provider
  • Hosted VoIP billing service provider
  • Internet Fax Service Provider
  • Internet Telephony Service Provider
  • Internet VoIP and Video Conferencing Service Provider
  • Marketing Services
  • Mobile VoIP
  • Network Service Provider
  • SIP Billing
  • VoIP Billing Software Provider
  • Voip consulting
  • Voip Termination ISP
  • Wholesale VoIP Carrier
  • Consultancy Services
  • Data Conferencing
  • E1 Bulk Wholeseller
  • E1/T1
  • Fax To Fax
  • Gatekeepering
  • gsm voip solutions
  • Hosted PBX
  • Hosted VoIP
  • Installation and Support Services
  • IP devices
  • Marketing Services
  • Outsourced Billing
  • Partnering for Origination and Termination
  • PC to Phone
  • PC to Phone, Phone To PC, Phone To Phone, Project Mangement Services, Termination, Web Call, Web To Phone
  • Phone To PC
  • Phone To Phone
  • Polycom
  • Project Mangement Services
  • SIP Softswitch & CPE
  • Sip Trunking
  • SIP VoIP Gateway
  • System Integration
  • Virtual PBX
  • Voice and Video Conferencing
  • Web Call
  • Web To Phone
  • 2N
  • Asterisk
  • Asterisk
  • Asterisk
  • Audiocodes
  • D-LINK
  • Digium
  • grandstream
  • Linksys
  • SIP softphones
Not data

Providers in database: 6430
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VoIP Favors SIP

VoIP Favors SIPSession Initiation Protocol (SIP) is a text-based standard put in place by the Internet Engineering Task Force (IETF) and is the primary protocol for Voice over Internet Protocol services. SIP is similar to a MGCP in that it's a signaling protocol that can create, maintain and terminate sessions in IP-based networks. These sessions include multimedia conferencing and two-way phone calls. SIP protocol can run on User Datagram Protocol (UDP), Stream Control Transmission Protocol (SCTP) and Transmission Control Protocol (TCP) because it was created to function independently from the underlying Transport Layer as an Application Layer.

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United States DgTec, LLC

Somos una operadora de telecomunicaciones. Conocidos como l

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