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On Wed, 6 Mar 2013, Paul Hirst wrote:
I can't actually get the voice menu to work when people phone my landline number. It doesn't recognise the tones. However when I install a SIP client on my mobile and connect that to the menu system in Asterisk it seems to work perfectly.You need to make sure that the SPA3000 and Asterisk are talking the same DTMF signalling protocol.I have it set to inband since apparently it doesn't correctly do any of the other options however even with inband Asterisk doesn't seem to detect it. I set up my extension to record the audio to a file and I can hear the tones in the file so it looks like inband is passing it through but Asterisk isn't detecting it. I couldn't find any options in Asterisk to detect the tones in the audio coming from a SIP extension, all the options seemed to be about how to transmit them. I hopefully correctly changed the audio codec to 64kbps ulaw, because as I understand it, otherwise they would be scrambled.
inband will only work the G711 u or a. Does it not support rfc2833 at all?Make sure your sip.conf entry for the spa3000 is the right runes in though. That's where you put the dtmf options.
I have read reports that the echo cancelation on the SPA3000 is a bit rubbish and that there is a serious issue with DTMF detection (I followed the instructions to solve it with no success so far). I have tried various firmware versions to try and fix this but nothing seems to have improved it.At least you haven't lost much money...That is a very positive way of looking at it! Hurrah :-)What I'd like to know is, should I just give up with the SPA3000, is it too old/rubbish? Is there any other way I can do this on the cheap? (Hopefully <£25). Have I given up too soon?You get what you pay for...I have had success with the Grandstream boxes, but even then, their echo cancellation hasn't been brilliant.Maybe I will jump ship to one of them. Hardware which comes vaguely recommended sounds like a better place than where I started from!I'm somewhat startled by how bad the SPA3000 seems to be but perhaps most people aren't routing it via Asterisk. I assume that if I was calling out via a Voip provider their echo cancelation would solve the problem?
Sometimes. The trick is to be digital for as long as possible. Preferably end to end. However whenever theres a 2-wire analog circuit involved (or the digital PSTN equivalent) then you'll have potential for echo.
Most people will probably connect an analog phone to the SPA3000. However I know someone in a call centre using SPA8000's to connect to existing analog phones which then connects to an Asterisk box ...
Similarly if I was allowing the SPA3000 to route the call from my landline directly to my DECT phone the latency might be low enough that the inbuilt echo cancelation would sort it out (or it would be low enough that my brain would sort it out)? However because I'm routing via asterisk I have introduced enough latency that neither can cope anymore. Does that explain how the SPA3000 can be so rubbish for my plans and yet apparently ok for other people?
If it's going direct from BT wall socket to a phone, then it should put a relay over the wires and not get in the signal path at all.
Most people will have ATAs like this plugged into their BT wall socket, talking to an external SIP provider with the analog port then going to the home analog phones. Some ATAs do allow SIP presentation of incoming analog calls to a SIP switch though - which is what you're doing and that's when the problems start... I've never had a successfull analog installation work like that - always used an asterisk box with an analog card.
Gordon
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