[asterisk-users] problem with outgoing calls

[mohamed ait tamghart]

hi,
firstly excuse me for my bad English
 I configured my astrerisk, and it goes for internal call but when I want to
make outgiong call I arriven't and the asterisk indicates the following
error





  == Using SIP RTP CoS mark 5
    -- Executing [0671735116 at default:1] Dial("SIP/100-0826a070", "SIP/
0671735116 at 10.76.252.3") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 0671735116 at 10.76.252.3
    -- Got SIP response 482 "Loop Detected" back from 0.0.0.0
    -- Now forwarding SIP/100-0826a070 to 'Local/0671735116 at default' (thanks
to SIP/10.76.252.3-08267f08)
    -- Executing [0671735116 at default:1] Dial("Local/0671735116 at default-6b02;2",
"SIP/0671735116 at 10.76.252.3") in new stack
[Jun  2 10:10:25] WARNING[6474]: app_dial.c:1437 dial_exec_full: Skipping
dialing interface 'SIP/0671735116 at 10.76.252.3' again since it has already
been dialed
  == Spawn extension (default, 0671735116, 1) exited non-zero on
'Local/0671735116 at default-6b02;2'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/100-0826a070' status is 'CHANUNAVAIL'





thanks for your help
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