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Summary:ASTERISK-20645: Outgoing Google Motif Calls connect but continue ringing on outgoing side
Reporter:Roy (coopvr)Labels:
Date Opened:2012-11-02 12:54:15Date Closed:2012-11-02 12:59:27
Priority:CriticalRegression?
Status:Closed/CompleteComponents:Channels/chan_motif
Versions:11.0.0 Frequency of
Occurrence		Frequent
Related
Issues:
Environment:Red Hat Linux 5Attachments:
Description: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@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
Comments:By: Roy (coopvr) 2012-11-02 12:59:08.655-0500

I forgot to mention I 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.

By: Joshua C. Colp (jcolp) 2012-11-02 12:59:27.173-0500

The issue tracker isn't the proper forum to get feedback on situations like this, it's for filing actual issues. I would suggest you move this question to the asterisk-users mailing list (I do monitor that list). If that conversation uncovers an actual issue one can be created here.
By: Roy (coopvr) 2012-11-02 13:01:40.352-0500

Ok how do I move it. I'm a new user here.
By: Joshua C. Colp (jcolp) 2012-11-02 13:04:16.338-0500

The asterisk-users is an email mailing list. You have to sign up at http://lists.digium.com/pipermail/asterisk-users/ and then can post.
By: Roy (coopvr) 2012-11-02 13:07:17.281-0500

Ok I did but there is an Issue with Outgoing Google Motif Calls connect but continue ringing on outgoing side.  I know others have or will have this problem to and need to be aware of it.  I have no idea if I fixed it or it just started working randomly.  I signed up and sent an email to the mailing list.  Thanks.