So I've got a bit of a head scratcher. Wanted to get some insight.
I've got a PBX running 12.3.0
We're a ULAW shop from end to end. But I've been playing with G722 just
for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom
IP650 (Same office).
Basically, Whenever I make an outbound call to a destination to something
not G722 ready, I get no incoming audio.
Example 1. From my T46G, I call the IP650. The call comes up in G722 and
we can both talk, No issues.
Example 2. From My T46G I call outbound to my cell phone. Our SIP trunk is
only G711, So The leg between the T46G and Asterisk is G722, And it has to
transcode it to G711. I get no receive audio on the T46G, However the
outbound audio is fine (I can hear my voice on the Cell phones receiver)
Example 3, I call inbound from my cell phone. Dial My extension. And
answer. Once again, The SIP Trunk leg is G711, And the T46G leg is G722, I
get audio both ways no problem.
Example 4. Exactly like Example 2, But I turn canreinvite= to yes. (I
normally use no, As I wish to always remain in the audio path). Same call
as Example 2 comes up, Quickly reinvites into G711 and audio works fine
both ways.
And finally, I do all of the above tests from the IP650, And everything
works fine. So it's limited to the T46G in some case.
My SIP config is below. Remember, I've toggled canreinvite=yes on and off.
But that's just a bandaid.
I also tried this on another T46G in the office running a much older
firmware and had the same issue...
[nick]
type=friend
context=flhsi-internal
secret=REDACTED
;insecure=yes
language=en
canreinvite=yes
host=dynamic
mailbox=106 at flhsi
notransfer=yes
dtmfmode=rfc2833
disallow=all
allow=g722
allow=ulaw
callerid="FLHSI Nick" <321-205-1100>
nat=no
call-limit=100
limitonpeer=yes
Nick Olsen
Network Operations (855) FLSPEED x106
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