My apologies I will clarify the situation. We set up Motif per Digium's new WIKI on Google Voice for Asterisk 11. It completed dialing / ring and answer BUT NO AUDIO.. No errors on the console.
We upgrade to SVN pull of Asterisk 11 and now Motif gives new errors (ICE). I gave up as there is little corresponding documentation on MOTIF. We rolled back to Asterisk 10 and got it to work within minutes using old GTALK/Jabber methodology. I know the rules about cross-post and before casting stones - I've been around Asterisk and other platforms for a long time. So thanks! On 10/10/12 2:45 PM, "Joshua Colp" <[email protected]> wrote: >Robert wrote: > >Hola, > >Please in the future don't cross post as you have done to both the >developer list and users list. If it's not related to development of >Asterisk the users list is where it should stay. > >> Just installed 11 and trying to get MOTIF / XMPP working to E164/PSTN >> number. >> >> We can get ring and a connected call but no audio >> >> SIP => ASTERISK => MOTIF >> >> Is there any specific configurations for getting audio to work? > >You haven't specified what you are calling out through, provided any >console log output, any network information, etc. This is really needed >before anyone can even come close to diagnosing your issue. > >To answer your question though there is nothing explicit you configure >to have audio work. It should "just work". > >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 > >-- >_____________________________________________________________________ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
