[Asterisk-Users] oh-323 / Cisco AS5300 problem

[Matias G.]

Leandro,
I don´t have the licenses on porpuose, I shouldn't need'em cause the phone I'm using has only g729 enabled, the oh323 has only g729a enabled and the Cisco (over which I  have no control at all) should be able to manage g729... furthermore reading the trace from the oh323 (debug level 5) I never found any reference to ulaw... just tenths of references to rtp audio being sent in g729 and then suddenly a:
51:45.239      ClearCallT...d:b092b248 H323    Clearing connection ip$localhost/32126 reason=EndedByLocalUser

on the second case, the phone is (99% sure) not busy (it's my desk phone) and if I call using g729 I get it to ring the phone and when I pick up I get the error, if I call using ulaw or alaw it doesn´t even ring I get the busy msg and that's it... but what is strange is that it can't be wrong rules in the 53xx cause when using g729 everything works fine till I answer the call... so the dialing logic is ok, of course I dial the same number when in g729 and g711...

any more clues?
thanks again.
M.

  ----- Original Message ----- 
  From: Leandro Tenorio 
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  Sent: Friday, June 03, 2005 3:32 PM
  Subject: RE: [Asterisk-Users] oh-323 / Cisco AS5300 problem



  The first error is probably because you don´t have licenses for 729 and you are trancoding the audio.

  The second is well dialed and you get from the 5300 a busy message, the reason could be user busy (as the message saids), wrong dial peer config, wrong dialing rules in 53xx.

  LTenorio



------------------------------------------------------------------------------
  From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matias G.
  Sent: Friday, June 03, 2005 1:02 PM
  To: asterisk-users at lists.digium.com
  Subject: [Asterisk-Users] oh-323 / Cisco AS5300 problem


  Hi i'm trying to connect to the PSTN  in the following way
  sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN

  I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 
  Asterisk-Oh323 0.7.2 pre1
  Open H323 v1.13.5
  pwlib v1.6.6

  and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and has a private ip but there's a stun server working and the connection works fine.

  If i try to dial out from the sip ATA through the GK (gnugk) configuring g729 in the sip account and the oh323.conf I get an error about trying to use ulaw:
     -- Executing Dial("SIP/D0103-3a43", "OH323/80152392522|50")
      -- H.323 call to 80152392522 with codec(s) g729
      -- Called 80152392522
      -- OH323/80152392522-ef38 is ringing
  Jun  3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to find 
  a path from ulaw to g729
  Jun  3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to find 
  a path from ulaw to g729
  OH323/80152392522-ef38: Format changed to ulaw (native g729).
  Jun  3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to find 
  a path from g729 to ulaw
  Jun  3 10:09:05 WARNING[27744]: app_dial.c:583 wait_for_answer: Unable 
  to forward voice
      -- Hungup 'OH323/80152392522-ef38'
    == No one is available to answer at this time (1:0/0/0)
      -- H.323 call 'ip$localhost/32125' cleared, reason 1 (Cleared by 
  local user)

  but if i try g711 ulaw on both the ata and the oh323 i get the following error
     -- Executing Dial("SIP/D0103-d16b", "OH323/00541152392522|50")
      -- H.323 call to 00541152392522 with codec(s) ulaw
      -- Called 00541152392522
      -- H.323 call 'ip$localhost/26583' cleared, reason 24 (Call ended 
  with Q.931 cause)
      -- OH323/00541152392522-d872 is busy
      -- Hungup 'OH323/00541152392522-d872'
    == Everyone is busy/congested at this time (1:1/0/0)

  I found a message about this error from January but there is no follow up

  any clues? 

  thanks a lot in advance.
  Matias


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