Any ideas? Still hoping..
On Sun, Apr 6, 2014 at 12:03 AM, Elliott W <dig...@private-address.info>wrote: > I have. > > On the receiving side I had gotten: > [2014-04-05 23:28:12] WARNING[1832] chan_iax2.c: Rejected connect attempt. > No secret present while force encrypt enabled. > > I had no secret because I was using RSA authentication and didn't think I > needed it, so I added EXACTLY the same line on both sides (copy/paste). > Now I get: > [2014-04-05 23:30:42] NOTICE[1832] chan_iax2.c: Call Terminated, Incoming > call is unencrypted while force encrypt is enabled. > > On the sending side I really get nothing useful: > [2014-04-05 23:30:42] VERBOSE[2795][C-00000002] pbx.c: -- Executing > [s@macro-dialout-trunk:22] Dial("SIP/comp-in-ch01-00000001", " > IAX2/ch01_ch02/1234,300,Ttr") in new stack > [2014-04-05 23:30:42] VERBOSE[2795][C-00000002] app_dial.c: -- Called > IAX2/ch01_ch02/1234 > [2014-04-05 23:30:43] VERBOSE[2795][C-00000002] chan_iax2.c: -- Hungup > 'IAX2/ch01_ch02-17634' > [2014-04-05 23:30:43] VERBOSE[2795][C-00000002] app_dial.c: == Everyone is > busy/congested at this time (1:0/0/1) > I modified the extension and the trunk name for security reasons, but > without force encryption calls flow back and forth easily. > > These three directives exist on both sides: > encryption=yes > forceencryption=yes > secret=mysecretcode > > So I'm kind of at a loss, I can see the options set, I can see: > [2014-04-05 23:59:32] VERBOSE[1832] chan_iax2.c: -- Accepting > AUTHENTICATED call from xxx.yyy.zzz.aaa: > when I DON'T have the force encryption set, so I can't see what else I > need to do.. > > CEW > > > > > On Fri, Apr 4, 2014 at 7:07 PM, Steve Totaro < > stot...@totarotechnologies.com> wrote: > >> Have you enabled IAX2 debugging and tried some test calls? >> >> Thanks, >> Steve T >> >> >> >> On Fri, Apr 4, 2014 at 6:59 PM, Elliott W <dig...@private-address.info>wrote: >> >>> That answered my question as to whether it WAS encrypted, I think, and >>> the answer is no, the credentials are but all the rest is not. That just >>> leaves the question of what I need to do to get it encrypted.. >>> >>> Thanks. >>> >>> >>> On Fri, Apr 4, 2014 at 12:59 PM, Steve Totaro < >>> stot...@totarotechnologies.com> wrote: >>> >>>> Wireshark. >>>> >>>> >>>> >>>> On Fri, Apr 4, 2014 at 11:13 AM, Elliott W <dig...@private-address.info >>>> > wrote: >>>> >>>>> Ok, I think I am 90%+ there. >>>>> >>>>> Note: the configuration or status is the same on both sides unless >>>>> otherwise noted. >>>>> >>>>> I am using RSA keys for authentication and the calls are coming >>>>> through as authenticated so I'm sure that part works. >>>>> >>>>> The peer shows the "(E)" next to the status in Asterisk Info for the >>>>> IAX2 peers >>>>> >>>>> The trunk configuration contains: >>>>> encryption=yes >>>>> >>>>> So here is my question, Calls stop flowing when I use the directive: >>>>> forceencryption=yes >>>>> At the trunk level or higher does not matter, same effect. >>>>> >>>>> So my question comes down to, are my calls getting encrypted and why >>>>> does this directive cause them to fail, AND how can I tell. >>>>> >>>>> Thanks. >>>>> >>>>> >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >>>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > >
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