Ok, problem solved
when using sjphone 1.60 for calling voxalot members.
I have found that the codec matching is extremely important as follows.
My SJphone is free version, which comes with audio codecs
GSM ( 8 kb/s plus overhead)
iLBC 30msec
iLBC 20 msec
G.711 A-law
G.711 U-Law
I am using 256 kb/s download and 64 kb/s upload
I had therefor concluded the best would be to use GSM and had disabled all the other codecs ( also with a view to check with a friend on Dialup at about 33 kb/s, hoping this would just sqeeze in understandable voise)
However the real testing was to another friend who runs a successful VoIP implementation using a Netcomm NB9W router and I think he now has a 1500/512 connection, ie he has no real audio issues using VoIP.
The modem default codec is G.729 and we believe it will respond to G.711's and I think G.723 but it does not have GSM capability and here is the problem.
My friend's voxalot number is recognised by the voxalot servers but also that the codecs can never match so I am passed to the voice service.
I think I proved this by enabling my sjphone with G.711 A and the phone now rings out first before I am then passed to voice service.
When using the sjphone log, I was beginning to be concerned about the statement:
Channel audio input: started
Capability: undefined
I think there is a clue here that the codecs do not match.
Conclusion is that if codecs don't match odd things can happen and it is not necessarily clear what the problem is.
well, in hind sight if I had purchased a registration including the g.729 I would never have spent many hours wading through different settings. On the positive side I know a bit more. Now to figure out why the xlite-3 has similar problems.