asterisk users - Nov 2012 - Outgoing Google Motif Calls connect but continue ringing on outgoing side

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
username=asterisk at gmail.com    ;;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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Roy Abshire wrote:
> 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.
Did you follow the guide at 
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google or just 
move the configuration files over and tweak them?
> 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.
So, you changed lots of settings and then it started working or did you 
give up after a failed call, come back, and it started working?

If it started working without any changes in between it could have been 
a temporary problem with the Google Voice gateway you were being 
connected to. I've seen this a few times during testing.
> 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.
<snip>
> Can someone tell me if these settings are correct? I have no idea but it
> works now.
Your settings seem fine.
> 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.
Yes, this is indeed a requirement.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

Roy Abshire wrote:
> I copied my settings over, and looked at the guide over and over to
> change the settings. But what the guide doesn't tell you is what you
> don't need anymore. So I didn't know if /Talk was ok or needed to
be
> omitted and externip or bindaddr was ok still because I had to have it
> for gtalk. I also don't know if transport=google or google-v1 is
> required and tried both.
The guide isn't written for migrations, it's for configuring from 
scratch. That's why. Treating it as a migration document may have caused 
stuff to go wonky.
> I kept tweaking sip, motif, and xmpp settings until it started working.
> I also tried reload at the console and finally started trying asterisk
> restart so I have no idea what helped.
Okay, so it sounds like something you did solved it. Without recreating 
the exact scenario and going through nothing stands out immediately.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

Roy Abshire wrote:
> I do have one thing I'm really unsure about.
>
> I'm using my Google Account for Asterisk and I'm also logged into
it
> from my Desktop Computer. Am I not supposed to be logged into this
> account and strictly use it for the Asterisk Server only? Does Asterisk
> have a problem knowing what Google Talk Login to use?
There's nothing explicit to prevent you from doing this but Google 
decides what client gets incoming calls, so it may go to your desktop 
when you don't want it to.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org