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@mific:1] Dial("SIP/4097-00000002",
"H323/[email protected]:1720,40") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Making call to [email protected]:1720 without gatekeeper.
Using 172.16.8.56 for outbound call
== New H.323 Connection created.
-- root is calling host [email protected]:1720
-- Call token is ip$localhost/19287
-- Call reference is 19287
-- DTMF Payload is 0x4235b48
-- Called [email protected]:1720
Setting capabilities to 0xc (ulaw|alaw)
Capabilities in preference order is (ulaw|alaw)
DTMF mode is 8
Allowed Codecs for ip$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@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
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