Trouble with Google Voice CID, and BYE response

I installed asterisk 13 from source on an old laptop and I have connected my gvoice account to it. Inbound/outbound calls are working but I have a couple issues.

  1. The incoming-motif CID is showing as the google voice number instead of the actual callers number

  2. When the incoming caller hangs up there appears to be no response from google telling asterisk to hang up the calls. It will just ring until timeout.

relevent extensions.conf:

[incoming-motif]

exten => s,1,NoOp(“Receiving call from google voice”)
same => n,Wait(1)
same => n,Answer()
same => n,Set(cnum=${CALLERID(number)})
same => n,Set(cnum=${cnum:2})
same => n,Set(CALLERID(all)="" <${cnum}>)
same => n,Dial(SIP/jeff&SIP/googletest,30,twD(:1))

[outgoing]
exten => _1XXXXXXXXXX,1,Dial(Motif/google/${EXTEN}@voice.google.com,r)

MOTIF.CONF
default
disallow=all
allow=ulaw
allow=h264
context=incoming-motif

google
transport=google-v1
context=incoming-motif
disallow=all
allow=ulaw
connection=google

XMPP.CONF

[google]
type=client
serverhost=talk.google.com
username=myusername@gmail.com
secret=mysecret
priority=25
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="I am available"
timeout=5

Here is the feedback from the CLI on a call (i replaced my gvoice number with 1234567890). I hang up after hearing the first ring from asterisk…

– Executing [s@incoming-motif:1] NoOp(“Motif/+1234567890-4eaf”, ““Receiving call from google voice””) in new stack
– Executing [s@incoming-motif:2] Wait(“Motif/+1234567890-4eaf”, “1”) in new stack
– Executing [s@incoming-motif:3] Answer(“Motif/+1234567890-4eaf”, “”) in new stack
– Executing [s@incoming-motif:4] Set(“Motif/+1234567890-4eaf”, “cnum=+1234567890”) in new stack
– Executing [s@incoming-motif:5] Set(“Motif/+1234567890-4eaf”, “cnum=234567890”) in new stack
– Executing [s@incoming-motif:6] Set(“Motif/+1234567890-4eaf”, “CALLERID(all)=”" <234567890>") in new stack
– Executing [s@incoming-motif:7] Dial(“Motif/+1234567890-4eaf”, “SIP/jeff&SIP/googletest,30,twD(:1)”) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
– Called SIP/jeff
– Called SIP/googletest
– SIP/jeff-00000000 is ringing
– SIP/googletest-00000001 is ringing
== Spawn extension (incoming-motif, s, 7) exited non-zero on ‘Motif/+1234567890-4eaf’

I’m not sure what other info is needed. I am using pjsip for NAT traversal

Can anyone point me in the right direction here?

I’m just revisiting this problem. Anyone have any ideas?