|Summary:||ASTERISK-20645: Outgoing Google Motif Calls connect but continue ringing on outgoing side|
|Date Opened:||2012-11-02 12:54:15||Date Closed:||2012-11-02 12:59:27|
|Environment:||Red Hat Linux 5||Attachments:|
|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.
;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
transport=google-v1 ;Using google or google-v1 didn't make a difference
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.
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.
I changed my sip settings for my google clients to:
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.
;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.
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.