VoIP hardware is developing fast - this is where you ask all those “how do I make my SIP Telephone, Adapter or Asterisk box work with my voip provider?” questions.
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By WelshPeter
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#5872
I want to be able to use my existing number for incoming calls but default to my VOIP service for outgoing. I have been told I am unable to port my number over, both by my VOIP provider and Plusnet my FTC supplier. The only solution that has been suggested is to use an adaptor with an FXO port, but there don't seem to be many available, and the ones that are cost a lot more money. The only one I've found is the Grandstream HT813. Will an FXO port do the job and are there any other alternatives?

Pete
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By WelshPaul
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#5873
Many people use their PSTN lines for incoming calls and a VoIP line for outgoing calls as it saves them money. For such setups its recommended that you implement the following:
  1. Have your VoIP provider set your PSTN landline number as its default caller id. Now, when you dial out over your VoIP line it looks like your calling from your PSTN line.
  2. Get yourself an analogue telephone adapter (or DECT base) that has both an FXO and FXS port. This will allow you to use a POTS and VoIP line together in perfect harmony. (e.g. answer calls that come in on your pots line and dial out over VoIP)
As you have found out, hardware is limited. I think it's because old copper lines are being phased out and should be obsolete by 2025 and so sales of such hardware are low. Anyway, I recommend the following hardware:

ATA's:
  1. Grandstream HT813 ATA
  2. OBIHAI OBi212 (available on amazon.com only)
DECT Bases:
  1. Gigaset N300 IP DECT
  2. Gigaset N300A IP DECT (as above but has its own built in answering machine)
You can of course look on eBay for older hardware such as the Cisco SPA232D, Linksys SPA3000 or LINKSYS SPA3102. Be careful though, most are probably going to be cheap Chinese knock-offs (I wouldn't risk it myself as its a security risk IMO).
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By WelshPeter
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#5877
Thank you Paul. I'll see if I can get an adaptor with FXO & FXS ports. I have a problem with my Cisco SPA 110 at the moment but I think it may be my VOIP provider. I answer a call but the line is silent. The caller just hears the ring tone as if the call has not been answered. The VOIP service from another provider works OK. A reboot sorts matters out but before long the issue returns. I'll contact them tomorrow to see if they can sort it.

Cheers
Pete
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By WelshPaul
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#5878
Try setting up the Stun option with the VoIP provider you're having issues with. See if they have their own, if not you should be able to use a STUN server offered by another provider.

If they don't have their own, try using Sipgate's:
  • STUN Server: stun.sipgate.net
  • STUN Server port: 3478
Good Luck
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By BrianG61UK
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#6184
WelshPaul wrote: Wed 8th Jul 2020, 09:54 ATA's:
  1. Grandstream HT813 ATA
  2. OBIHAI OBi212 (available on amazon.com only)
I couldn't make this work satisfactorily with the Grandstream HT813.
Calls coming in on the FXO port from the PSTN line would ring on the phone connected to the FXS port but occasionally when I answered the phone it would not connect me to the call, instead just giving me a dial tone ready to call on via VoIP. This was definitely before the incoming call had gone away, I know because while testing I had another phone connected directly to the PSTN line and that one was still ringing. Also the HT813 has a facility where you can have a dial plan and direct certain dialled numbers to the PSTN line (use can use for such things as 999, 1471, 17070, and perhaps 0800). But that was no good because it dialled out on the PSTN line and then waited a few seconds before allowing you to hear the audio from the line. You would miss the beginning of announcements from answering machines or the first digit as 17070 reads back you number.
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