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@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
>
>
>
>
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