Hi guys! Help please - all our incoming calls have a limit on 60 seconds - i dont know why - can you help me?
all internal/external peers are online and pinging
everything seems good
but call fails after 60 seconds - why?
- Sometime carrier have a limited on the length of a call if no RTP packets is received. Make sure you have two way audio.
- Check NAT Rules to make sure the firewall is not closing the connection.
Just some ideas.
I have mostly nothing on rtp.conf:
rtpstart=10000
rtpend=20000
I have 2 way audio - during this 60 seconds everybody hears each other
Nat has no problems - i have this effect on all 20 users in our crm system !
Can you help me?
You need to provide the logging from Asterisk. At the moment we don’t even know which side cleared the call/
Sixty seconds is of the right order of magnitude for a missing ACK.
Also, you haven’t indicated the version of Asterisk that you are using. WebRTC is a moving target and you need to always use the very latest version.
Now i switch to 17.5.1 and making a new tests
sip.conf
session-timers=refuse
make everything ok with incoming calls - why? may be i should change some other settings?
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