I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and
jabber.conf to use motif.conf and xmpp.conf.
I disabled gtalk and jabber from loading in modules.conf
noload => res_jabber.so
noload => chan_gtalk.so
After copying my settings to the new conf files and restarting Asterisk
I had no errors, but making outgoing calls from clients just kept
ringing even though the other side picks up and hears nothing.
I played with my settings for days and have no idea what I changed that
got it working so I'm hoping someone can tell me what caused this and
maybe what I changed that fixed it.
Now it works but I don't know why so I'd like some feedback.
My Asterisk Server is NOT behind a NAT but my Clients are and I'm using
Google Voice for incoming and outgoing calls.
Here is what I have done.
I completely removed my [general] section from motif.conf and added a
[default](!) and transport=google-v1 like the example states. The
[general] section was needed in gtalk.conf to get it working but seems
to not be of any use now.
[general]
;context=incoming ;;Context to dump call into
;bindaddr=0.0.0.0 ;;Address to bind to
;bindaddr=76.12.113.228
;externip=76.12.113.228
;disallow=all
;allow=ulaw
;allowguest=yes ;;Allow calls from people not in peer list
[default](!)
disallow=all
allow=alaw
allow=ulaw
allow=h264
transport=google-v1 ;Using google or google-v1 didn't make a difference
context=incoming
[asterisk](default)
connection=asterisk
I removed the /Talk suffix from my xmpp.conf username fields and changed
timeout=5. It took me a while to notice the /Talk was not needed anymore.
[asterisk]
type=client ;;Client or Component connection
serverhost=talk.google.com ;;Route to server for example,
talk.google.com
[email protected] ;;Username with optional resource.
secret=secret ;;Password
priority=1 ;;Resource priority
port=5222 ;;Port to use defaults to 5222
usetls=yes ;;Use tls or not
usesasl=yes ;;Use sasl or not
status=available ;;One of: chat, available, away,
xaway, or dnd
statusmessage="Asterisk Server" ;;Have custom status message for
Asterisk.
timeout=5
I changed my sip settings for my google clients to:
[asterisk]
host=dynamic
type=friend
nat=force_rport,comedia
canrevinvite=no
qualify=yes
dtmfmode=rfc2833
context=home
disallow=all
allow=ulaw;h263
Can someone tell me if these settings are correct? I have no idea but
it works now.
I also made sure port 5060 and 5222 was open in iptables
I also had to change rtp.conf to add icesupport=yes. I use my own rtp
port range that is opened on the firewall.
[general]
icesupport=yes
rtpstart=15000
rtpend=20000
;rtpchecksums=no
;dtmftimeout=3000
;rtcpinterval = 5000 ; Milliseconds between rtcp reports
; strictrtp=yes
I also had to add icesupport=no in sip.conf[general]section to stop
"failed to extend" errors happening for SIP calls.
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