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[jleed at me.com]

Hey guys,
have issues with reinvite, no matter what endpoint is calling asterisk always tries switch simple_bridge to native_rtp
Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge technology to native_rtp


in endpoints table “direct_media” sets to “no” on all endpoints but it doesn’t help.

if native_rtp not work for some reason I have oneway audio. how can I fix this? if I add mix_monitor it works, but it’s not a right way to fix this issues.

Asterisk 13.2.0

[mjordan at digium.com]

On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed@me.com> wrote:

[jleed at me.com]

Well, it breaks audio for all NAT endpoints, how can I fix this?

[mjordan at digium.com]

On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <jleed@me.com> wrote:

[jleed at me.com]

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:99@192.168.1.73 (99@192.168.1.73):5060,20)
-- Called PJSIP/99/sip:99@192.168.1.73 (99@192.168.1.73):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>

[mjordan at digium.com]

On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <jleed@me.com> wrote:

[jleed at me.com]

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