[asterisk-users] Problem H323 asterisk 1.6.2.19

troxlinux xserverlinux at gmail.com
Wed Jul 27 13:32:00 CDT 2011


Hi list ,  I am connecting one avaya with asterisk by h323 and when I
call to avaya becomes disconnected, this  is my debug


ippbx*CLI> h323 set debug on
H.323 Debugging Enabled
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Executing [1083 at mific:1] Dial("SIP/4097-00000002",
"H323/1083 at 172.16.8.5:1720,40") in new stack
    -- Requested transfer capability: 0x00 - SPEECH
 -- Making call to 1083 at 172.16.8.5:1720 without gatekeeper.
Using 172.16.8.56 for outbound call
        == New H.323 Connection created.
        -- root is calling host 1083 at 172.16.8.5:1720
        -- Call token is ip$localhost/19287
        -- Call reference is 19287
        -- DTMF Payload is 0x4235b48
    -- Called 1083 at 172.16.8.5:1720
Setting capabilities to 0xc (ulaw|alaw)
Capabilities in preference order is (ulaw|alaw)
DTMF mode is 8
Allowed Codecs for ip$localhost/19287 (ip$172.16.8.56:39935):
         Table:
   G.711-uLaw-64k <1>
   G.711-ALaw-64k <2>
   UserInput/hookflash <3>
   UserInput/basicString <4>
 Set:
   0:
     0:
       G.711-uLaw-64k <1>
       G.711-ALaw-64k <2>
     1:
       UserInput/hookflash <3>
     2:
       UserInput/basicString <4>

        -- Sending SETUP message
        -- Received RELEASE COMPLETE message...
        -- ClearCall: Request to clear call with token
ip$localhost/19287, cause EndedByRemoteBusy
        -- Sending RELEASE COMPLETE
        ExternalRTPChannel Destroyed
        ExternalRTPChannel Destroyed
        ExternalRTPChannel Destroyed
        ExternalRTPChannel Destroyed
        -- ClearCall: Request to clear call with token
ip$localhost/19287, cause EndedByTransportFail
-- 1083 was busy
        == H.323 Connection deleted.
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [1083 at mific:2] Hangup("SIP/4097-00000002", "") in new stack
  == Spawn extension (mific, 1083, 2) exited non-zero on 'SIP/4097-00000002'


I have perfectly compiled h323 in asterisk

core show channeltypes
Type        Description                              Devicestate
Indications  Transfer
----------  -----------                              -----------
-----------  --------
Local       Local Proxy Channel Driver               yes          yes
        no
Bridge      Bridge Interaction Channel               no           no
        no
H323        The NuFone Network's Open H.323 Channel  no           yes
        no
Console     OSS Console Channel Driver               no           yes
        no
USTM        UNISTIM Channel Driver                   no           yes
        no
Phone       Standard Linux Telephony API Driver      no           yes
        no


any idea?

regardss




-- 
rickygm

http://gnuforever.homelinux.com



More information about the asterisk-users mailing list