[Asterisk-video] Patch 0010217

Valerio Puglia valerio at oscorp.sm
Mon Mar 17 05:16:58 CDT 2008 

Klaus tnx for response..
i try your dialplan but not work the called thelephone swith to video 
and remain in waiting......and i hear in the audio the negotation...In 
the caller thelephone is in waiting....without video and audio.
i attach the log


Klaus Darilion ha scritto:
> Hi Valerio!
>
> Your dialplan is wrong. You have two choices:
>
> 1. Forward incoming call without decoding video. That means asterisk 
> will forward the digital data from one call to the other call. H324M 
> negotiation is end-2-end between the mobile phones. There are 2 ISDN 
> calls, but logically only one H324M session.
>
> [from-pstn]
> exten => 1,1,Set(CHANNEL(transfercapability)=VIDEO)
> exten => 1,2,NoOp(transfer=${CHANNEL(transfercapability)})
> exten => 1,3,Set(CHANNEL(userinformationlayer1)=38)
> exten => 1,4,NoOp(ul1=${CHANNEL(userinformationlayer1)})
> exten => 1,5,Dial(Zap/0043123456)
>
> 2. Forward call with decoding/encoding video. That means, that h324m_gw 
> will decode the H324M session into Asterisk audio and video frames. 
> Thus, for the outgoing call leg you need h324m_call to encode the frames 
> again. Thus, there are again 2 ISDN call, but this time we have 
> logically 2 H324M session. The first from caller to h324m_gw and the 
> second from h324m_call to the callee. Make sure to set the 
> transfercapability  just before the outgoing Dial command.
>
> [from-pstn]
> exten => _X.,1,h324m_call(${EXTEN}@h324m-decoded)
>
> [h324m-decoded]
> exten => _X.,1,h324m_call(${EXTEN}@h324m-decoded-encoded)
>
> [h324m-decoded-encoded]
> exten => _X.,1,Set(CHANNEL(transfercapability)=VIDEO)
> exten => _X.,2,NoOp(transfer=${CHANNEL(transfercapability)})
> exten => _X.,3,Set(CHANNEL(userinformationlayer1)=38)
> exten => _X.,4,NoOp(ul1=${CHANNEL(userinformationlayer1)})
> exten => _X.,5,Dial(Zap/0043123456)
>
> Hope that works. Please report your results.
>
> klaus
>
> Valerio Puglia schrieb:
>   
>> 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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>>>>>
>>>>>   
>>>>>       
>>>>>           
>>>>     
>>>>         
>>> _______________________________________________
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>>>
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>>>
>>>   
>>>       
>>     
>
> _______________________________________________
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>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-video
>
>   


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

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