Quote:
Originally Posted by boatman
I have only IP phones here. The only reason I wanted to dial the gateway was to test that it worked before giving the number to someone near that gateway.
|
But that's the problem with testing from a VoIP phone. I would suggest making the test call from either a friend's "land line", or from a cell phone.
Quote:
Originally Posted by boatman
I use the G711u codec.
|
Then give "InBand" a try, and see how it works for you. While InBand is technically the least reliable of the options, it's also the option that is most universally supported (assuming you use the G711u CODEC).
BTW: What "InBand" really does is send the tones as if they were just some other "voice" on your VoIP line. This means that if your VoIP can properly send this "voice" sufficiently undistorted (which is why the less distorting, but high bandwidth, G711u CODEC is needed), than the tones WILL get to the other end no matter what tone support is available on the other end. However, for the same reason this is the least reliable (albeit the most universally "supported") of the options, as anything that distorts the tonal quality of your "voice" will distort the tones (and cause problems with them on the other end).
Quote:
Originally Posted by boatman
I would like to study up on what all those different DTMF settings mean, but I can't find the technical docs for the SPA2102, Linksys seems to have abbreviated the user manual for this model. If anyone can point me to the full docs for the SPA2102, that would be super.
|
If you are willing to go with an older PDF manual for a slightly different (but closely related) model adapter, check out the downloadable manual on the Sipura web site (these adapters used to be made by Sipura, before LinkSys bought them out):
http://www.sipura.com/Documents/Sipu...uidev2.0.9.pdf
http://www.sipura.com/support/index.htm
BTW: From the above listed manual, here is the short versions of what the DTMF settings mean/do:
Method to transmit DTMF signals to the far end:
Inband = Send DTMF using the audio path;
INFO = Use the SIP INFO method,
AVT = Send DTMF as AVT events;
Auto = Use Inband or AVT based on outcome of codec negotiation
FYI: I used to have my adapter set to "Auto", but changed it to "InBand" after discovering that my primary VoIP provider's voice mail didn't work with "Auto" (but did with InBand explicitly forced). Go figure...