I have now fixed this issue, and am posting this for the benefit of anyone else
who may be suffering with a similar problem.
It was, as I suspected all along, a subtle misconfiguration at this end.
The fix was to give the SIP trunk its own configuration stanza in sip.conf as
follows;
[sip_trunk_outbound]
type=peer
host=provider.sld.cc
disallow=all
allow=alaw
and replace all instances of
Dial(SIP/provider.sld.cc/44${EXTEN:1})
with
Dial(SIP/sip_trunk_outbound/44${EXTEN:1})
In the absence of that important little stanza, the [general] settings were
applying to the ad-hoc SIP endpoint; meaning that even in spite of explicitly
setting the outbound SIP codec, Asterisk was insisting to use G726.
No sooner had I worked this out, than the SIP trunk provider e-mailed
basically to confirm my thinking.
The moral of this story: Never trust a configuration file written by someone
else, no matter how close it was to working ;)
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
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