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 mjor...@digium.com wrote:
On Thu, Mar 19, 2015 at 1:47 AM, Nick
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/304-0022, /pbx/agi.php) in
new stack
-- Launched AGI Script /pbx/agi.php
-- AGI Script Executing Application: (Dial)
On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome jl...@me.com 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/304-0022, /pbx/agi.php) in
new stack
-- Launched
On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome jl...@me.com wrote:
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
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
On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome jl...@me.com wrote:
Well, it breaks audio for all NAT endpoints, how can I fix this?
Local (packet to packet) bridging should not do that. Remote (direct
media) can do that.
Can you confirm - by looking at a verbose level 4 log - how Asterisk
is
Well, it breaks audio for all NAT endpoints, how can I fix this?
On 18 Mar 2015, at 15:48, Matthew Jordan mjor...@digium.com wrote:
On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome jl...@me.com
mailto:jl...@me.com wrote:
Hey guys,
have issues with reinvite, no matter what endpoint is
