How to Extend Connection to A Remote Site

March 10, 2016 by Ying-Hui Chen

Our tech support team at VoIP Supply offers great pre and post-sales support plus provisioning, consultations, configuration, and installation help.

We get a lot of VoIP hardware and software questions and would like to share the solutions with everyone. Here is a new real question and answer from VoIP users like you.

 

HOW TO EXTEND CONNECTION TO A REMOTE SITE

CaptureQ: I have a remote site that has a network but is not connected to the internet. The site is an island 12 miles from shore. I have a wireless network that connects the remote island to shore. What I need is something that I can extend a POTS telephone line across the wireless network to the island, as we have need for telephone out there. We do not have access to the internet. Is it possible to build a system that can do this?

A: We can extend that connection with two Grandstream ATAs and Peer to Peer HandyTone Scenario (Extend Analog Lines Using the HandyTone 503 without SIP Server). grandstream ata

Grandstream HT502 ATA

The Grandstream HT502 is based on the SIP 2.0 standard and features 2 FXS ports, dual 10M/100Mbps Ethernet ports with integrated high performance NAT router, port status and message waiting LED, and a base stand for vertical positioning. Learn more here.

Grandstream HT503

Grandstream HT503 is a hybrid Analog Telephone Adapter and VoIP Router featuring both an FXS (analog telephone) port and an FXO (PSTN) Port so you can have backup lifeline support using the PSTN in the case of a power outage. It also integrates 2 10/100 RJ45 ports with an integrated high performance NAT router. Learn more here.

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4 Comments

  • Yuesendel Garcia

    Hi. I’m from Venezuela and really don’t know if this is the right place to ask. I have both HT503 and HT801, I can page or dial from HT801 and it will ring on the HT503 but once it’s answered there’s no audio on either end. ISPs on both sides use NAT so I can’t make a direct connection. Tried same configuration using LAN and got same problem. Is there a workaround to connect them using some SIP free service?

    • Hi Yuesendel,

      Thank you for your question! You have to port forward UDP 5060 for SIP/SDP and 10,000-20,000 for RTP/Audio packets OR you have a NAT issue. Also, ensure you’re not using any SIP ALG or SIP sessions in your router/firewall. VoIP Supply works with multiple SIP trunk hosting providers. Contact our VoIP experts at 1-800-398-8647 today to get more information today! If you have more questions, you are welcome to submit a ticket here: https://voipsupply.zendesk.com/hc/en-us

      Thank you,
      Evy

  • Rudi

    Hi,

    I need to setup the same type of setup as above, with 17x PSTN lines on an old analogue PBX in one building, and a new Asterisk server with the same 17 extensions in another building.
    The Asterisk server will also add additional extensions to the 2nd building, and the additional extensions in the 2nd building need to be able to dial the existing 17 existions.

    How to proceed?

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