No Audio Problems


#1

I’ve got an Openser+Asterisk server going behind a firewall. My clients are all behind nats too.

Phone-to-Phone calls work great. I used this url to set up asterisk: openser.org/dokuwiki/doku.php?id … ysql_views

Asterisk has no audio though. My question is:

  1. How do I discover what part of the audio delivery is failing?
  2. What is a recommended codec for linux/win32? I’m using x-lite.

Here’s some debug so I know the call is getting to asterisk:

3(14009) REQUEST: INVITE, sip:*86@68.121.238.19, 68.121.238.21
4(14011) REQUEST: INVITE, sip:*86@68.121.238.19, 68.121.238.21
4(14011) Invite: Call to check voicemail
4(14011) *** uri != myself
– Executing VoiceMailMain(“SIP/68.121.238.19-081c9830”, “mpapet%406812123819”) in new stack
– Playing ‘vm-login’ (language ‘en’)
– Playing ‘vm-password’ (language ‘en’)
– Incorrect password ‘’ for user ‘mpapet%406812123819’ (context = default)
– Playing ‘vm-incorrect-mailbox’ (language ‘en’)
– Playing ‘vm-password’ (language ‘en’)
– Incorrect password ‘’ for user ‘mpapet%406812123819’ (context = default)
– Playing ‘vm-incorrect-mailbox’ (language ‘en’)
– Playing ‘vm-password’ (language ‘en’)
– Incorrect password ‘’ for user ‘mpapet%406812123819’ (context = default)
– Playing ‘vm-incorrect’ (language ‘en’)
– Playing ‘vm-goodbye’ (language ‘en’)
– Executing Hangup(“SIP/68.121.238.19-081c9830”, “”) in new stack