On 05/03/2016 at 05:43 PM Joshua Colp wrote: > Michael Maier wrote: >> On 05/03/2016 at 04:50 PM Joshua Colp wrote: >>> Michael Maier wrote: >>>> Hello Joshua! >>>> >>>> >>>> I attached the sip debug without the progressinband=never set. The >>>> caller didn't get a ring back tone as expected. >>> Please keep this on list so that anyone who may run into a similar >>> problem in the future has a chance of finding this discussion. >> >> You are right - normally I'm going exactly this way. But I don't want >> the traces to be world wide readable (-> privacy). I will write a >> summary to the list as far as we know more. >> >>> As for your log there's nothing of note really, it's just expecting to >>> send the ringing as inband audio instead of out of band. Does "rtp set >>> debug on" show the RTP traffic going to the other side? >> >> Yes. I attached it. >> >> And no - there isn't any packet blocked by iptables :-). > > There is nothing abnormal here and Asterisk appears to be doing the > correct thing. It's sending an audio stream with early progress to the > caller. It may be that in a previous FreePBX, or when used with 13, they > changed the behavior for this to force early media and the provider is > not allowing it. >
Ok - but this doesn't seem to answer my main question: Why must progressinband=never be applied especially if asterisk uses it by default? The big difference between w/ and w/o it is: w/o the option progrssinband=never just the sip-package 183 Session Progress is sent. w/ the option set, the additional sip-packages 100 Trying 180 Ringing 180 Ringing are sent. If progrssinband=never is the default, the Ringing package should be sent always, shouldn't it? If I remove the option progrssinband=never via FreePBX, I can't find any other value provided to progrssinband in /etc/asterisk/*. Why does it work always correctly w/ the second trunk, which is connected directly to the extension? Is it possible to switch off the standard behavior of asterisk / progrssinband for ring groups only by setting some other options? Thanks, kind regards, Michael -- _____________________________________________________________________ -- 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