I'm having a problem with a new SIP trunk.
Calls within the UK to fixed lines are fine, but calls to mobiles have
noticeably poorer audio quality.
I thought it might have been a codec issue; we have used G.726 for internal
and external calls (over primary ISDN and GSM). So I tried allowing "alaw",
(G.711 A-law) which is the native codec used within the PSTN in this country,
but this made no improvement.
We had
disallow=all
allow=g726
in the [general] section of sip.conf. In the section for one of the phones, I
added
allow=alaw
and then inserted
Set(SIP_CODEC=alaw)
in the relevant part of extensions.conf. For good measure, I also added
NoOp(Codec was ${SIP_CODEC})
in the "h" extension. The messages in the Asterisk CLI appeared to show that
the audio codec was correctly being set to "alaw", and on hangup I got "Codec
was alaw", but there was no improvement to the sound quality.
Is there something I am doing wrong, or do I need to get in touch with our SIP
trunk provider?
--
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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