I have asterisk 1.4.1 version and in console I see SIP information like
-- Executing [303@office:1] Dial("SIP/itc-gianluca-0070d0d0", "SIP/503@bel") in new stack
-- Called 503@bel
-- SIP/bel-0072eef0 is ringing
-- Call on SIP/bel-0072eef0 left from hold
-- SIP/bel-0072eef0 answered SIP/itc-gianluca-0070d0d0
-- Packet2Packet bridging SIP/itc-gianluca-0070d0d0 and SIP/bellinzona-0072eef0
== Spawn extension (office, 503, 1) exited non-zero on 'SIP/itc-gianluca-0070d0d0'
-- Executing [204@from-ribo:1] Dial("SIP/vivacom-0072c9f0", "SIP/itc-erika") in new stack
-- Called itc-erika
-- SIP/itc-erika-007267e0 is ringing
-- SIP/itc-erika-007267e0 is ringing
-- SIP/itc-erika-007267e0 is ringing
-- SIP/itc-erika-007267e0 is ringing
And this is fine, I can see all trafic, requests, etc…
Recently I’ve installed 2 new asterisk v1.4.25.1 and I have almost empty console. I can’t se these info. What is the problem? Which settings I need to change?
with regards