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.
;context=incoming ;;Context to dump call into
;bindaddr=0.0.0.0 ;;Address to bind to
;allowguest=yes ;;Allow calls from people not in peer list
disallow=all allow=alaw allow=ulaw allow=h264
transport=google-v1 ;Using google or google-v1 didn’t make a difference context=incoming
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.
type=client ;;Client or Component connection serverhost=talk.google.com ;;Route to server for example, talk.google.com firstname.lastname@example.org ;;Username with optional resource. secret=secret ;;Password priority=1 ;;Resource priority portR22 ;;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:
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.
;rtcpinterval = 5000 ; Milliseconds between rtcp reports
I also had to add icesupport=no in sip.conf[general]section to stop
“failed to extend” errors happening for SIP calls.