Problem H323 asterisk 1.6.2.19
Tags: call, Driver, lt 1, lt 2, networking issue, sip, UserInput, vpn tunnels
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″,
> “H323/1083@172.16.8.5:1720,40″) in new stack
> — Requested transfer capability: 0×00 – SPEECH
> — Making call to 1083@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@172.16.8.5:1720
> — Call token is ip$localhost/19287
> — Call reference is 19287
> — DTMF Payload is 0x4235b48
> — Called 1083@172.16.8.5:1720
> Setting capabilities to 0xc (ulaw|alaw)
> Capabilities in preference order is (ulaw|alaw)
> DTMF mode is 8
> Allowed Codecs for ip$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@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
>
>
>
>