[Asterisk-Users] H323 call dropped when answered

[Krystian.Filiks]

Hi,
This is the scenario
I have the SJlabs phone with g711ulaw active and the rest disabled.
I have * with chan_h323
I have a Quintum DX  that supports, g723.1 , g729AB, ulaw and alaw.

The problem is that, it does not mather what I put in the 
extensions.conf  I have tried all possible ways that I so far could find 
using the net.
I tried all possible codecs ulaw, alaw, g723 and g729 always the same 
result.
The phone rings but as soon as answered it dissconnects.

The debug shows

    -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack

    -- Called 0797617729

    -- H323/0797617729 is ringing

    -- H323/0797617729 answered SIP/sj1-4ff7

  == Spawn extension (default, 0797617729, 1) exited non-zero on 
'SIP/sj1-4ff7'

    -- Executing Dial("SIP/sj1-4ff7", "H323/h") in new stack

    -- Called h

  == Spawn extension (default, h, 1) exited non-zero on 'SIP/sj1-4ff7'

 The first Dial is normal but the 2^nd Dial  "Executing 
Dial("SIP/sj1-4ff7", "H323/h") in new stack"

Where do that come from?

PLEASE someone HELP!

The * have the config below

In extensions.conf I use
[globals]
[default]
exten => _.,1,Dial(H323/${EXTEN})
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
in H323.conf I use
[general]
port = 1720
bindaddr = 195.216.65.212
tos=lowdelay
allow=all
gatekeeper = 195.216.65.215
AllowGKRouted = yes
context=default
[AST37]
type=h323
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

In SIP.conf I have
[general]
port=5060                   
bindaddr=xxx.xxx.xxx.xxx

[sj1]
type=friend                
context=default
host=dynamic        
disallow=all            
allow=all            
username=sj1                
secret=sj1

[sj2]
type=friend                   
context=default
host=xxx.xxx.xxx.xxx          
allow=ulaw                  
username=sj1           
secret=sj1
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

administrator tootai wrote:

> Krystian Filiks a écrit :
>
>> Like you suggested I tried the g.711 now and got the same, The called 
>> number rings but when answered it dropped.
>> I connect to a Quintum Tenor DX.
>>
>> The part I'm curious about is  6:53.985           
>> Transactor:8140ee8    h323trans.cxx(678)   Trans   admissionRequest 
>> rejected: requestDenied
>>  6:53.988          H225 Caller:8159198         h323.cxx(2660)  
>> H225    Gatekeeper refused admission: requestDenied
>>  6:53.959          H225 Caller:813c890      h323pdu.cxx(1159)  
>> H225    Read error (0):
>>
>> Does anyone have a clue where to look for the problem?
>>
>> here is a trace,
>> -- Executing Dial("SIP/sj1-a7e9", "H323/h at 195.216.65.215") in new stack
>> Allowed Codecs:
>>         Table:
>>   G.711-uLaw-64k{sw} <1>
>> Set:
>>   0:
>>     0:
>>       G.711-uLaw-64k{sw} <1>
>>
>> -- Making call to h at 195.216.65.215 using gatekeeper.
>>                channelsOpen = 1
>>                channelsOpen = 0
>>  6:53.959          H225 Caller:813c890      h323pdu.cxx(1159)  
>> H225    Read error (0):
>>        == New H.323 Connection created.
>>        -- sj1 is calling host h at 195.216.65.215
>>        -- Call token is ip$localhost/31767
>>        -- Call reference is 31767
>>    -- Called h at 195.216.65.215
>>        -- ClearCall: Request to clear call with token ip$localhost/31767
>>        -- Sending RELEASE COMPLETE
>>  == Spawn extension (default, h, 1) exited non-zero on 'SIP/sj1-a7e9'
>>  6:53.985           Transactor:8140ee8    h323trans.cxx(678)   
>> Trans   admissionRequest rejected: requestDenied
>>  6:53.988          H225 Caller:8159198         h323.cxx(2660)  
>> H225    Gatekeeper refused admission: requestDenied
>>  6:54.004                 H323 Cleaner         h323.cxx(1542)  
>> H323    Connection ip$localhost/31766 terminated.
>> -- Call with Tenor Gateway [195.216.65.215] completed (EndedByLocalUser)
>>        == H.323 Connection deleted.
>>  
>>
> What's h at 195.216.65.215? If you are register to GK H323/<EndPoint> is 
> enough. I don't understand your h EP. And also request denied seems 
> that you need to register. But I don't know how work Quintum, maybe 
> I'm wrong.
>
>> -----Original Message-----
>> From: asterisk-users-admin at lists.digium.com 
>> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of 
>> administrator tootai
>> Sent: Thursday, August 12, 2004 6:02 PM
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [Asterisk-Users] H323 call dropped when answered
>>
>> Krystian.Filiks a écrit :
>>
>>  
>>
>>> Hello anyone that can help me here?? please read below.
>>> [...]
>>>
>>>   
>>>
>>>> Allowed Codecs:
>>>>
>>>>         Table:
>>>>
>>>>   G.723.1{sw} <1>
>>>>
>>>> Set:
>>>>
>>>>   0:
>>>>
>>>>     0:
>>>>
>>>>       G.723.1{sw} <1>
>>>>
>>>>     
>>>
>> G.723.1 is not a codec in * Use g711 instead. If your GK is GnuGK, 
>> see his debug logs. Also, run asteriks in debug mode and check logs 
>> in full file.
>>
>>  
>>
>
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