I wonder under what conditions asterisk switches from simple_bridge to native_rtp and doing route optimization?
I have 2 nodes with exactly the the same asterisk configs.
Only difference is that they are connected to different SIP-trunks (2 different voip providers)
One does route optimization and the other one does’nt.
Both endpoint config:
direct_media : true
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
Thanks in advance,
Same codec, same packetization, no features that need audio, DTMF method the same. Those are the general ones. If you enable debug in logger.conf it’ll tell you why it’s not allowing it when the decision is made.
Thanks, great, I will check and get back with result here for future generations
I found it!!!
Very simple error, I had recording activated on one channel.
Could see it when executing
pjsip show channel xxxx
for both channels.
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