Trying Asterisk On AWS
I am trying to get audio to work on AWS using asterisk 18.14.0
I have enabled the firewall to allow ALL UDP on AWS
My SIP extension has nat=force_rport,comedia qualify=yes allow=ulaw allow=alaw allow=gsm canreinvite=yes
I enable “rtp set debug on” and the console is printing info.
The call comes into my linphone softphone – but I get no audio on my linphone softphone. What might I be missing to allow the audio ?
Volume is up.
Thanks
Jerry
8 thoughts on - Trying Asterisk On AWS
I just noticed the RTP log is sending to 192.168.2.0 which is my local lan address of the linphone – it should be sending to the NAT address and is not. What did I not set correctly ?
I am not using pjsip – but the older asterisk.
Thanks
Jerry
Have you configured chan_sip to know it is behind NAT itself and what its public IP address is? If not, then you’ll get no audio.
>Have you configured chan_sip to know it is behind NAT itself and what its
I’m thinking I have not. What did I miss ?
Thanks,
Jerry
The sample configuration file outlines how things work, and the options for it:
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874
in general localnet and externip (or externaddr, or externhost)
I added:
externip=xxx nat=force_rport,comedia
to the general section of sip.conf
its still sending to the local IP.
Jerry
Look at the actual SIP/SDP signaling to see what is being sent and what IP
addresses are being used. If they’re correct, then see if you are receiving traffic using “rtp set debug on”. If you aren’t then it’s something outside of Asterisk preventing incoming traffic. Until Asterisk receives traffic it can’t know the IP address+port to send outgoing to, beyond what was given in the SIP/SDP.
Does your local router (the one connecting Linphone to the Internet) have a
“SIP helper” or “SIP ALG” feature?
If so, ensure that it is turned off.
Antony.
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Thanks for the information This is now working…
externip