On 05/03/2016 at 09:16 PM Joshua Colp wrote: > Eric Wieling wrote: >> I don't know the default setting for progressinband in the code, but it >> is documented in Asterisk 11's sip.conf.sample as defaulting to never. >> Maybe the docs were fixed since Asterisk 11. > > The behavior change to actually do what the option was documented to do. > As part of that the default was changed to reflect the past behavior, > thus why it was changed to no. The commit itself: > > chan_sip: make progressinband default to no > > After the "progressinband" value setting of "never" was updated to never > send a 183 this separated its use from the "no" value.
But "never" option therefore sends 180 Ringing which I was missing. The new default "no" doesn't send 180 Ringing any more ... . > Since "never" was > the default, but most users probably expect "no" this patch updates the > default for the "progressinband" setting to "no." > > This was tracked under ASTERISK-24835[1]. > > [1] https://issues.asterisk.org/jira/browse/ASTERISK-24835 > This makes sense! I migrated from asterisk11-11.8.1-40_centos6.x86_64, which had the default progressinband=never to asterisk13-core-13.7.2-1.shmz65.1.94.x86_64 which had the new default. POTS callers advertise support for early media - mobile callers on the other hand don't advertise it, therefore mobile wasn't a problem because early media (183) isn't triggered (and used!) at all. Two strange things being left: 1. Why does progressinband=no work, if there is *no* ringgroup between trunk and extension. This seems to be a "feature" of FreePBX. 2. Why is early media used even if the caller doesn't advertise it? Are there other triggers like P-Early-Media? Another basic question: What do I need early media exactly for? I'm only using SIP phones - nothing else. Couldn't it be completely disabled for these trunks? Or would it break things like voice mail service e.g.? How can I disable it completely even if it is advertised by the caller? Thanks, 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