[Asterisk-Users] oh323 channel : transport failure

[M. Ehsanul Karim]

Hello,

Trying to call a h323 enpoint  Sjphone and NetMeeting  each of the I
get a meesage trasport fail . I tried with all codec but same results.
Can you please help ?

Executing SetGlobalVar("SIP/2001-bfec", "OH323_OUTCODEC=alaw") in new stack
    -- Setting global variable 'OH323_OUTCODEC' to 'alaw'
    -- Executing Dial("SIP/2001-bfec",
"OH323/88029880144 at 216.236.116.251") in new stack
    -- H.323 call to 88029880144 at 216.236.116.251 with codec ALAW
    -- Called 88029880144 at 216.236.116.251
    -- OH323/L32532 is ringing
    -- H.323 call 'ip$localhost/32532' cleared, reason 8 (Transport failure)
    -- Hungup 'OH323/L32532'
  == No one is available to answer at this time
Oct  8 08:18:17 WARNING[1294355]: pbx.c:1933 ast_pbx_run: Timeout, but
no rule 't' in context 'from-sip'
    -- Executing SetGlobalVar("SIP/2001-045d", "OH323_OUTCODEC=alaw")
in new stack
    -- Setting global variable 'OH323_OUTCODEC' to 'alaw'
    -- Executing Dial("SIP/2001-045d",
"OH323/88029880144 at 216.236.116.251") in new stack
    -- H.323 call to 88029880144 at 216.236.116.251 with codec ALAW
    -- Called 88029880144 at 216.236.116.251
    -- OH323/L32533 is ringing
    -- H.323 call 'ip$localhost/32533' cleared, reason 8 (Transport failure)
    -- Hungup 'OH323/L32533'
  == No one is available to answer at this time