[Asterisk-video] Patch 0010217

[Valerio Puglia]

Hi Klaus



> Valerio Puglia schrieb:
>   
>> hi Klaus
>>
>> i remove AST_FORMAT_ULAW and it works
>>     
>
> what does work?
>   
the problem of the calling phone didn't listen the answer
after cancell AST_FORMAT_ULAW the caller  is ok




>> but when bridge 2 mobile thelephone or call from sipphone to meobile 
>> phone the video doesn't start.....
>>     
i try to use asterisk to bridge 2 mobilecall after the call is 
established the call is hungup

mobilephone > asterisk >mobilephone


[from-pstn]


exten => _x.,1,h324m_call(666 at video-out2)


[video-out2]
exten => 666,1,Set(CHANNEL(transfercapability)=VIDEO)
exten => 666,2,NoOp(transfer=${CHANNEL(transfercapability)})
exten => 666,3,Set(CHANNEL(userinformationlayer1)=38)
exten => 666,4,NoOp(ul1=${CHANNEL(userinformationlayer1)})
exten => 666,5,Goto(call2,666,1)


[call2]
exten => 666,1,Dial(Zap/g0/xxxxxxxxxx)







Spawn extension (call2, 666, 1) exited non-zero on 
'Local/666 at video-out2-f169,2'
    -- Channel 0/22, span 4 got hangup request, cause 16
    -- Hungup 'Zap/94-1'
    -- Hungup 'Zap/115-1'
    -- Accepting call from 'xxxxxx' to 'xxxx' on channel 0/23, span 4
    -- Executing [xxxx at from-pstn:1] h324m_call("Zap/116-1", 
"666 at video-out2") in new stack
    -- Executing [666 at video-out2:1] Set("Local/666 at video-out2-45df,2", 
"CHANNEL(transfercapability)=VIDEO") in new stack
    -- Executing [666 at video-out2:2] NoOp("Local/666 at video-out2-45df,2", 
"transfer=VIDEO") in new stack
    -- Executing [666 at video-out2:3] Set("Local/666 at video-out2-45df,2", 
"CHANNEL(userinformationlayer1)=38") in new stack
    -- Executing [666 at video-out2:4] NoOp("Local/666 at video-out2-45df,2", 
"ul1=38") in new stack
    -- Executing [666 at video-out2:5] Goto("Local/666 at video-out2-45df,2", 
"call2|666|1") in new stack
    -- Goto (call2,666,1)
    -- Executing [666 at call2:1] Dial("Local/666 at video-out2-45df,2", 
"Zap/g0/xxxxx") in new stack
    -- digital call, setting user information layer 1 to 38 (0x26)
    -- zap call: h324musellc=0, ast->userinformationlayer1=38
    -- Requested transfer capability: 0x18 - VIDEO
    -- Called g0/3468442617
    -- Zap/94-1 is ringing
    -- Zap/94-1 answered Local/666 at video-out2-45df,2
  == Spawn extension (call2, 666, 1) exited non-zero on 
'Local/666 at video-out2-45df,2'
    -- Channel 0/23, span 4 got hangup request, cause 16
    -- Hungup 'Zap/94-1'
    -- Hungup 'Zap/116-1
















>
> I do not understand - above your write that it works now?
> klaus
>   
>>
>>
>>
>> Klaus Darilion ha scritto:
>>     
>>> Hi Valerio!
>>>
>>> I guess it as a codec problem inside asterisk. app_h324m tunnels the 
>>> digital call inside G711. Then, sometimes asterisk tries to transcode 
>>> from alaw to ulaw.
>>>
>>> Please search in app_h324m.c for AST_FORMAT_ULAW and remove it (there is 
>>> some comments which tell you how to do it), so that h324m_call forces 
>>> the usage of ALAW (which is the default of zaptel when using E1).
>>>
>>> Let me know if this worked for you.
>>>
>>> regards
>>> klaus
>>>
>>> Valerio Puglia wrote:
>>>   
>>>       
>>>> Hi Klaus i have inserted your patch in asterisk 1.4.17 and libpri.. the 
>>>> call out work prefect but when arrive the videocall ..and i accept the 
>>>> call the telephone remaing in wait (also is resond) but the sip phone 
>>>> the call is already upcoming ... asterisk doesn't listen the answer...
>>>> i try to SIPPHONE > TO CELL
>>>> and bridge 2 mobile phone... but the same result...the celluallar phone 
>>>> caller remain to calling state...but the other is waiing for video(like 
>>>> as it had answered)
>>>>
>>>> do you have any idea for my problem?
>>>>
>>>>     
>>>>         
>>> _______________________________________________
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>>>
>>>   
>>>       
>>     
>
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>   


-- 
Best regards,
Valerio Puglia
OScorp S.P.A.
NETWORK Adm