[asterisk-bugs] [JIRA] (ASTERISK-20645) Outgoing Google Motif Calls connect but continue ringing on outgoing side

[Roy (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20645?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=199189#comment-199189 ] 

Roy edited comment on ASTERISK-20645 at 11/2/12 1:00 PM:
---------------------------------------------------------

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.

                
      was (Author: coopvr):
    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

                  
> Outgoing Google Motif Calls connect but continue ringing on outgoing side
> -------------------------------------------------------------------------
>
>                 Key: ASTERISK-20645
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20645
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_motif
>    Affects Versions: 11.0.0
>         Environment: Red Hat Linux 5
>            Reporter: Roy
>            Severity: Critical
>
> 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

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