[asterisk-users] Problem H323 asterisk 1.6.2.19

[Vladimir Mikhelson]

Do you have any network devices or VPN tunnels in between the Asterisk
and Avaya?

The reason I am asking it looks like a potential networking issue.

Has this setup ever worked before?

-Vladimir




On 7/27/2011 1:32 PM, troxlinux wrote:
> 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
>
>
>
>