Ok, if this is normal why I have oneway audio when nat endpoint calling to local. if mixmonitor or srtp is enabled audio is ok. Issues with native_rtp for sure
Sent from my iPhone > On 19 Mar 2015, at 23:08, Matthew Jordan <[email protected]> wrote: > >> On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <[email protected]> wrote: >> NAT endpoint calling local endpount - switching to native_rtp then no audio, >> both of them have direct_media=no, Verbose log: >> >> -- Executing [99@dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in >> new stack >> -- Launched AGI Script /pbx/agi.php >> -- AGI Script Executing Application: (Dial) Options: >> (PJSIP/99/sip:[email protected]:5060,20) >> -- Called PJSIP/99/sip:[email protected]:5060 >> -- PJSIP/99-00000023 is ringing >> -- PJSIP/99-00000023 answered PJSIP/304-00000022 >> -- Channel PJSIP/304-00000022 joined 'simple_bridge' basic-bridge >> <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28> >> -- Channel PJSIP/99-00000023 joined 'simple_bridge' basic-bridge >> <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28> >>> Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from >> simple_bridge technology to native_rtp >>> Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in >> stack >>> Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in >> stack >>> 0x7f4b50145420 -- Probation passed - setting RTP source address to >> 194.204.157.200:8972 >>> 0x7f4b5014f140 -- Probation passed - setting RTP source address to >> 192.168.1.73:5004 >> -- Channel PJSIP/304-00000022 left 'native_rtp' basic-bridge >> <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28> >> -- Channel PJSIP/99-00000023 left 'native_rtp' basic-bridge >> <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28> >> -- <PJSIP/304-00000022>AGI Script /pbx/agi.php completed, returning 4 > > Correct - and per the log, they shouldn't be in a direct media bridge: > >> Locally RTP bridged 'PJSIP/99-00000023' and > 'PJSIP/304-00000022' in stack >> Locally RTP bridged 'PJSIP/99-00000023' and > 'PJSIP/304-00000022' in stack > > Locally RTP bridged means media is still flowing through Asterisk, it > just isn't being decoded and passed through the core. > > > -- > Matthew Jordan > Digium, Inc. | Director of Technology > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://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
