Hi,
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 bmdemo.asterisk@gmail.com (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
res_xmpp.so Asterisk XMPP Interface 0
1 modules loaded
ubuntuCLI> module show like motif
Module Description Use Count
chan_motif.so Motif Jingle Channel Driver 0
1 modules loaded
Here is my dialplan:
[color=#BF4080]
[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/bmdemo.asterisk@gmail.com,r) [pbx_config]
2. Hangup() [pbx_config]
’_1XXXXXXXXXX’ => 1. Dial(Motif/google/${EXTEN}@voice.google.com,r) [pbx_config]
2. Hangup() [pbx_config]
Here is my motif.conf file:
default
disallow=all
allow=ulaw
allow=h264
context=incoming-motif ; Default context that incoming sessions will land in
;
;
; Added by MLA per instruction wiki.asterisk.org/wiki/display/A … ing+Google
;
[google] (default)
transport=google
connection=google
Here is my xmpp.conf file:
ubuntu*CLI> !cat /etc/asterisk/xmpp.conf
[general]
debug=yes ; Enable debugging (disabled by default).
[google]
type=client
serverhost=talk.google.com
username=bmdemo.asterisk@gmail.com
secret=REDACTED
priority=25
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="Yo yo yo"
timeout=5