[asterisk-users] Only 8kHz recorded after disallowing all but G722 codec on inbound

[David P]

We want to record inbound channels at 16kHz, but send only 8kHz to our
peers. I've set our default profile in sip.conf to disallow all but g722,
and the peers disallow all but ulaw. We have a proxy in front of Asterisk
that is configured to disallow all but G722 also.

My test calls show inbound to the proxy is recorded at 16kHz, inbound in
Asterisk is only 8kHz, and the peers receive 8kHz. So the only thing not
working is Asterisk's sampling rate on inbound, and it seems to be
downsampling.

After a lot of web searching, I can't find any explanation of why we're not
getting 16kHz for G722. We're using Asterisk 14.7.6.

Cheers,
David
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Re: [asterisk-users] How to ignore REFER entirely with chan_sip or PJSIP ?

[Daniel Tryba]

On Fri, Jun 15, 2018 at 05:32:30PM +0200, Olivier wrote:
> In my testing, I saw that Asterisk always included a REFER value in each
> INVITE's Allow header, no matter how allowtransfer/allow_tranfer was set.
> 
> Is there a way to remove this REFER value entirely either globally or
> specifically for a given peer/endpoint ?

No, not with asterisk itself.

> Which telephony feature would loose without REFER method ?

AFAIK none except the ability to REFER (obviously).

What you can do is:
-route calls via a proxy that gives you the ability to modify SIP on
the fly (e.g. kamailio)
-wihh chan_sip use disallowed_methods (= REFER)
-with pjsip reject a REFER in the dialplan:
 https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers

The last 2 options will make PBXs/endpoints that use REFERs, since
asterisk is advertising it, fail in certain scenarios. So cleanest is
the first option, which will take some work redesigning your setup but
might be a good thing on the long run.



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