No - but this is a new setup so I can't say it worked before...it just isn't working from the start.
I've found the call setup works and once bridged there is one way audio (to the ATA, none from the ATA). And the the connection drops after 30 secs approx because something on the path (or endpoint) realizes something is wrong... ________________________________ From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Davis [stda...@multiservice.com] Sent: Sunday, December 15, 2013 12:41 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Did you change your network switch recently? Some Digium IAX ATAs do not behave well with Cisco equipment. On Sat, Dec 14, 2013 at 10:26 PM, Michelle Dupuis <mdup...@ocg.ca<mailto:mdup...@ocg.ca>> wrote: meant to say restart didn't help either.. ________________________________________ From: asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com> [asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of Michelle Dupuis [mdup...@ocg.ca<mailto:mdup...@ocg.ca>] Sent: Saturday, December 14, 2013 11:20 PM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Ok just restart -----Original Message----- From: asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com> [mailto:asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of Michelle Dupuis Sent: Friday, December 13, 2013 11:46 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing I tried transfer=no, transfer=yer, and transfer=mediaonly (with a "reload" inbetween)....same result I agree it sounds like something either end is using the wrong IP/port address somewhere in the call (yet signalling works fine). Anything else to suggest? I was hoping for an externalip type setting but not in iax2 (at least not in 1.4.x.x) ________________________________________ From: asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com> [asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of Joshua Colp [jc...@digium.com<mailto:jc...@digium.com>] Sent: Friday, December 13, 2013 11:44 AM To: Asterisk Users List Subject: Re: [asterisk-users] IAX2 bridge failing Michelle Dupuis wrote: > Some more details...I noticed that the call is bridged, and audio goes > one way. However, the dial command still times out after 35 seconds > (approx), and exists non-zero. > While the channels are up, I did an core show channel xxx and found > Blocking in: > ast_waitfor_nandfds > Is this a bug? Or something I can fix through config? Hola, Set "transfer=no" under the entries in iax.conf for the peers/users/friends/etc in question, reload, retry, and see if that changes the behavior. If it does then something involved may not like IAX2 native transfers. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com<http://www.digium.com> & www.asterisk.org<http://www.asterisk.org> -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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 -- Steven Davis VoIP Engineer Multi Service +1-913-663-9748 o +1-913-871-5155 m stda...@multiservice.com<mailto:stda...@multiservice.com> [http://www.multiservice.com/assets/images/logos/ms_email_no_tagline.png]<http://www.multiservice.com/> ------------------------------------------------------------------ This email is intended solely for the use of the addressee and may contain information that is confidential, proprietary, or both. If you receive this email in error please immediately notify the sender and delete the email.. ------------------------------------------------------------------
-- _____________________________________________________________________ -- 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
