[asterisk-users] *SOLVED* SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

[A J Stiles]

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 .