In my last topic Joe wrote that my audio problems are probably nat-related. I can say for sure (ok, 99%), that it is not the case.
I cannot hear that:
Quote:
-- <SIP/616XXX-00000000> Playing 'digits/4.gsm' (language 'en')
-- Playing 'point' (escape_digits=) (sample_offset 0)
-- <SIP/616XXX-00000000> Playing 'digits/4.gsm' (language 'en')
-- <SIP/616XXX-00000000> Playing 'digits/2.gsm' (language 'en')
-- Playing 'cents-per-minute' (escape_digits=) (sample_offset 0)
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/616XXX-00000000> Playing 'digits/1.gsm' (language 'en')
-- <SIP/616XXX-00000000> Playing 'digits/thousand.gsm' (language 'en')
-- <SIP/616XXX-00000000> Playing 'digits/1.gsm' (language 'en')
-- <SIP/616XXX-00000000> Playing 'digits/hundred.gsm' (language 'en')
-- <SIP/616XXX-00000000> Playing 'digits/30.gsm' (language 'en')
-- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
-- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
-- <SIP/616XXX-00000000> Playing 'digits/3.gsm' (language 'en')
-- Playing 'prepaid-seconds' (escape_digits=#) (sample_offset 0)
I have GSM codec available in my sip client. I have sounds in folders: /var/lib/asterisk/sounds/en/digits/ AND in /var/lib/asterisk/sounds/digits/en/. I also had them in /var/lib/asterisk/sounds/digits/ but it still did not work. Chown is set to asterisk:asterisk and chmod to 755 for all files. Right after that when it starts connecting through the trunk, I can hear audio and during the call I can also hear "You have 60 seconds left".
Why?