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


 
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PostPosted: Thu Jul 31, 2014 10:49 am    Post subject: [asterisk-users] *SOLVED* SIP trunk gives fuzzy / distorted Reply with quote

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 Wink


--
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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