Steve. Thanks for the insight. I won't pretend to know what "early-audio" is, but I guess I'm about to find out :-).
Also, I believe that I have a nearly identical setup like this with the exact same SIP provider w/o any trouble. However, I think that system must be running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to confirm). Is there a significant difference between 1.2/1.4 & 1.6 in this scenario? Thanks a million!! :-) - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . -----Original Message----- From: Steve Davies [mailto:[email protected]] Sent: Thursday, April 07, 2011 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No ringback even though progressinband=yes is set On 7 April 2011 17:02, Douglas Mortensen <[email protected]> wrote: > Any ideas on why callers who call into my customer's SIP trunk are not > hearing a ringback tone? I had this on one other asterisk system, and wound > up needing to set progressinband=yes in the SIP trunk config. > > I have set this on the current system & restarted asterisk, but to no avail. > > I am using: > > AsteriskNOW distro > Asterisk build is 1.6 from AsteriskNOW repository: > asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 > > Any help would be greatly appreciated! :-) > > - > Doug Mortensen In my personal experience with SIP and 1.6.x, that mostly depends on where you are sending the call to. It depends on whether the next or subsequent leg tries to use early-audio for the ring tone, or uses a Ringing event to signal that is what is happening. It then depends on whether the originating caller's equipment can understand early-audio ringing. We have a setup here where all our trunks support early-audio ringing except one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure ringing occurs. Perhaps provide more details? Or you may find that tracing the SIP gives you the clue that you need. Hope that helps, Steve -- _____________________________________________________________________ -- 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
