Aastra 9112i with tb 2.8.0.4 no audio on outgoing calls, incoming are OK

Trying to configure tb 2.8.0.4 with Aastra 9112i phones.

When I first set up the phones tb was not "seeing" them as being registered, however, I was able to make outgoing calls just fine.

In my efforts to get the phones registered, I managed to get tb to see them and the phones were able to call each other and with audio both directions. Everything's cool. Then I noticed that only incoming calls were working as expected. Outgoing calls now have no audio either way. I don't believe this is limited to a config issue with the phones as incoming calls routed to custom extensions (e.g. LOCAL/14055551212@outbound-allroutes) have no audio either.

My best guess is that this may be a codec issue.

My router settings mirror that of my two other working tb installs.
My sip_nat.conf file mirrors that of my other two working tb installs.

What I am needing is a nudge in the right direction on what to check next. Like I said, incoming calls are fine. Outgoing calls have no audio either way.

I'm not sure which config details I should post but I am happy to post any/all config details needed to sort this out.

Here's my aastra.cfg:

#===================================================
# "aastra.cfg" file

alert emergency: 3
alert group: 1
alert internal: 2
alert priority: 4
date format: 7
dhcp: 1
dndkey value: 0

http server: 0.0.0.0
http path:
ring tone: 2

sip dial plan terminator: 0
sip dial plan: "[2-9]11|[56]1[1-5]|958|8502|[2-9]xxxxxx|1[2-9]xxxxxxxxx|*1[2-9]xxxxxxxxx|011x+#|xx*|*xx+#|x+#"
sip digit timeout: 5
sip dtmf method: 2
sip nortel nat support: 1
sip outbound proxy port: 0
sip outbound proxy: 0.0.0.0
sip proxy ip: 192.168.1.88
sip proxy port: 5060
sip registrar ip: 192.168.1.88
sip registrar port: 5060
sip backup proxy ip: xxxx.xxxxxxx.xxx - this has a legit server defined
sip backup proxy port: 5060
sip backup registrar ip: xxxx.xxxxxxx.xxx - this has a legit server defined
sip backup registrar port: 5060
sip registration period: 3600
sip rtp port: 3000
sip session timer: 0
sip transport protocol: 1
time format: 1
time server1: 129.6.15.28
time server2: tick.uh.edu
time server3: server 1.us.pool.ntp.org
time zone code: CST
time zone minutes: 0
time zone name: US-Central
#===================================================

Here's one of my extension .cfg files:

#===================================================
# "MAC.cfg" file

displayName1: "Kris Smith x201"
############################
displayName2: "My Company Name"
sip screen name: "Kris Smith"
sip user name: 201
sip line1 auth name: 201
sip line1 password: xxxxxxxxxxxxxxxx
sip line1 user name: 201
#===================================================

What am I missing?