Caller ID
curt00:
The demo at http://www.obihai.com/uidemos/obi200/i has the same format that I have on my OBi202, not the format in your screenshot.
I have tried making calls from my Bell Canada POTS line, cell phone and Google Voice to my SIP number assigned to SP2 (PH1). CallerID is not displayed on my Nortel analog phone.
I went to Physical Interfaces/Phone1 Port/Port Settings and CallerIDMethod is FSK(Bell202) (greyed out) and CallerIDTrigger is After First Ring (greyed out).
I welcome any other ideas or suggestions.
SteveInWA:
Here's a screenshot from my OBi 202, in the USA, and a screenshot of the call history.
The one difference is that my call history has a "1" digit before the area code in every case. That could be messing up your phone's ability to read the CID data. Your next step is to try a different/better SIP ITSP, like Canadian company voip.ms, and a different analog telephone.
curt00:
My Physical Interfaces/PHONE1 Port settings look the same as your screenshot, except that my ChannelTxGain has a value of -2. Does this make a difference?
You are correct. My Call History looks like your screenshot, except that you have the "1" digit before the area code. It seems strange that a missing "1" digit would mess up the CID. You would think that the system should simply display whatever numbers there are, that are inside the round brackets.
It would seem like a lot of work to figure out how voip.ms works and to port my 3 numbers over from freephoneline.ca. I took a quick look at voip.ms' website and it is not intuitive. Have you heard much about voip.ms and freephoneline.ca? Is voip.ms much better than freephoneline.ca?
SteveInWA:
Channel transmit gain is the amplification level of the signal that is sent from the OBi to your telephone's earpiece (in other words, it makes the other party's voice louder in your ear). Since the phone is still on-hook during the ring period, it isn't applicable to your situation. I just have mine cranked up a bit because I have a mild hearing loss.
I'm not suggesting that you take drastic action and port your numbers over at this time. I'm suggesting that you obtain a new number from voip.ms and see if their service works for you. If you care about CID, then it's up to you to decide if spending a few bucks to troubleshoot it is worthwhile to you. You can always cancel it if it doesn't work out.
In my opinion, and that of many others on this forum, voip.ms is one of the highest-quality ITSPs, and the fact that they're a Canadian company suggests that they are experienced with making everything work correctly for Canadian customers. I've never read a report of CID not working with voip.ms.
My motto: nothing's "free" -- you don't get what you don't pay for...in this case "freephoneline"
curt00:
SteveInWA:
I'm not against paying a little, but it's the time. I looked at voip.ms and it looks like it will be another project for me to take on. I don't work in the phone industry and it has taken already many hours to figure out the acronyms, read setup guides and to get things to work with Freephoneline. I fear that it will take many more hours to get things to work with another VoIP provider.
I have a few analog phone numbers that I use occasionally. I found out about Freephoneline as a way to save money. I didn't think that I would need to do so much to get things to work. I do not want to risk taking on another project by taking on another phone company. I will probably live without the Caller ID.
Thanks for your help. I really appreciate it.
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