Hello!
I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2. As customer of German Telekom, I have three numbers and therefore three trunks - each number is bound to one trunk. After migration, some callers complained about missing ringback tone: they didn't hear any ring tone and where surprised that they suddenly got me anyway :-). The connection afterwards was as expected. Deeper investigation yielded, that a few caller groups have been affected by this problem: - All POTS customers of German Telekom - VoIP customer of T-Systems (usually companies which transfered their telephone system) Not affected have been callers like All-IP customers of German Telekom or any tested mobile caller or caller using other telecommunications companies. To make it even more strange: Calls coming from T-Systems customer via call forwarding have been working fine, too. And: The ringback tone wasn't missing, if the second number (the second trunk) of the asterisk installation was used! The only difference between those two trunks is: The first trunk is configured to a ring group - the second trunk is configured directly to an extension. My solution after long time of digging around: I added progressinband=never to sip_general_additional.conf But this solution confuses me, because http://www.voip-info.org/wiki/view/Asterisk+sip+progressinband tells: progressinband=never Whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behaviour of Asterisk. ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Why do I have to provide it especially if it is the default behavior? Why did it work without this option with asterisk 11? Why is there suddenly a difference in behavior between binding a trunk to a ring group or an extension? Puzzled, regards, Michael Maier -- _____________________________________________________________________ -- 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
