Pls, what could be wrong?

photonspear
Joined: Wed 16 of Feb, 2011

Pls, what could be wrong?

Posted:Wed 16 of Feb, 2011 (18:34 UTC)
I just built an asterisk server using asterisk 1.8.2 version, following step-by-step documentation; along the line I created some SIP accounts & corresponding dialplan extensions. Effort was made to test the configuration by initiating phone calls; expected display (in line with documentation) was shown on the asterisk server CLI but the two parties involved could not hear their conversation though they could heard the ringing tone. I used idefisk 2.0 softphone on SIP channel. Pls, what could be wrong? And suggestion on way forward; could this be NAT issue?. Here below is the display on the asterisk CLI
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Using SIP RTP CoS mark 5
    -- Executing [2018@users:1] Dial("SIP/ict1-00000009", "SIP/ict2,20") in new stack
  == Using SIP RTP CoS mark 5
    -- Called ict2
    -- SIP/ict2-0000000a is ringing
    -- SIP/ict2-0000000a answered SIP/ict1-00000009
    -- Remotely bridging SIP/ict1-00000009 and SIP/ict2-0000000a
  == Spawn extension (users, 2018, 1) exited non-zero on 'SIP/ict1-00000009'
  == Using SIP RTP CoS mark 5
    -- Executing [2017@users:1] Dial("SIP/ict2-0000000b", "SIP/ict1,20") in new stack
  == Using SIP RTP CoS mark 5
    -- Called ict1
    -- SIP/ict1-0000000c is ringing
    -- SIP/ict1-0000000c answered SIP/ict2-0000000b
    -- Remotely bridging SIP/ict2-0000000b and SIP/ict1-0000000c
  == Spawn extension (users, 2017, 1) exited non-zero on 'SIP/ict2-0000000b'
localhost*CLI>

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