Trouble getting Asterisk 11 to make outgoing GoogleTalk call


I have set Asterisk and successfully made SIP calls between two endpoints. I am running into trouble with getting a SIP endpoint to communicate with GTalk. I have been attempting this for over a week with no luck and after doing a fair bit of RTFM online and in this forum, I am at a loss as to why I continue to experience this error:

[Jun 29 22:35:29] WARNING[1639][C-00000000] app_dial.c: Unable to create channel of type ‘Motif’ (cause 42 - Switching equipment congestion)
[Jun 29 22:35:29] VERBOSE[1639][C-00000000] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[Jun 29 22:35:29] DEBUG[1639][C-00000000] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.

when I attempt to place a call from a Jitsi client running on my native MacOS to (logged in via Google Talk app on my Android) through an Asterisk 11.0 running on Ubuntu in a VMWare VM on the same laptop.

I know there is something I am fundamentally misunderstanding about the way this is supposed to work, so I have included the relevant bits of configuration below. I would be grateful for any insight into what I might be doing wrong.

I know for a fact that res_xmpp and chan_motif are compiled and loaded:

ubuntuCLI> module show like xmpp
Module Description Use Count Asterisk XMPP Interface 0
1 modules loaded
CLI> module show like motif
Module Description Use Count Motif Jingle Channel Driver 0
1 modules loaded

Here is my dialplan:
[b]ubuntu*CLI> dialplan show
[ Context ‘incoming-motif’ created by ‘pbx_config’ ]
‘s’ => 1. NoOp() [pbx_config]
2. Wait(1) [pbx_config]
3. Answer() [pbx_config]
4. SendDTMF(1) [pbx_config]

[ Context ‘DPlan1’ created by ‘pbx_config’ ]
‘100’ => 1. Dial(SIP/smp2) [pbx_config]
‘200’ => 1. Answer() [pbx_config]
2. Playback(hello-world) [pbx_config]
3. Hangup() [pbx_config]
‘300’ => 1. Dial(Motif/google/,r) [pbx_config]
2. Hangup() [pbx_config]
’_1XXXXXXXXXX’ => 1. Dial(Motif/google/${EXTEN},r) [pbx_config]
2. Hangup() [pbx_config]

Here is my motif.conf file:

context=incoming-motif ; Default context that incoming sessions will land in
; Added by MLA per instruction … ing+Google
[google] (default)

Here is my xmpp.conf file:

ubuntu*CLI> !cat /etc/asterisk/xmpp.conf
debug=yes ; Enable debugging (disabled by default).

statusmessage="Yo yo yo"

The way I use google talk is to have it call the other party and forward that to me. Meaning that google calls me and connects me to the other party.

I don’t remember reading where google talk acts like a SIP provider, but maybe you are doing something I don’t know about.

And let’s say I picked up my phone and dialed, this thing connected to google voice and sent it the number to dial and then called me back on my phone, but I still have the phone off hook, so does it somehow transfer me to the new incoming call?