On Sat, Jun 15, 2013 at 4:03 PM, Daniel Tryba <dan...@tryba.nl> wrote: > On Sat, Jun 15, 2013 at 03:02:41PM -0400, Andre Goree wrote: >> Setting the CID did not work, unfortunately :( > [...] >> I'm going to try another number that we have through them in hopes >> that it'll complete and I'll let you know if that works. Do you have >> any other suggestions on what you think they might be filtering by? >> >> In the trap given to me by the company, they show our system issuing a >> "disconnect" from our end, rather than their end dropping the call. > > Do a "pri set debug" (or whatever it is called in 1.4 (zap?)) Zap/Zap > bridging should work, it did on my PRIs and still does with DAHDI. Only > thing I can think of is the TON/NPI might be a problem (but doubt it > since SIP/Zap works). > >
Thanks so much for your suggestions. I'm running 1.0.x (yes, archaic, and in fact my actual task is migrating this system to asterisk11+Freepbx -- very fun in and of itself without regards to this issue...but I digress), and so I needed to run "pri debug span <span>", which I've now done. I attempted the call again have pasted the debug output here: http://pastebin.com/cHHnMfh6 I can't thank you enough for your assistance, and I understand if you wouldn't want to go through the debug output as it's LONG -- though I'm thinking most of the pertinent info as towards the end. -- _____________________________________________________________________ -- 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