Summary: | ASTERISK-20645: Outgoing Google Motif Calls connect but continue ringing on outgoing side | ||
Reporter: | Roy (coopvr) | Labels: | |
Date Opened: | 2012-11-02 12:54:15 | Date Closed: | 2012-11-02 12:59:27 |
Priority: | Critical | Regression? | |
Status: | Closed/Complete | Components: | Channels/chan_motif |
Versions: | 11.0.0 | Frequency of Occurrence | Frequent |
Related Issues: | |||
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. [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. |