[Asterisk-video] Patch 0010217

[Klaus Darilion]

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?
>>>>>
>>>>>     
>>>>>         
>>>> _______________________________________________
>>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>>
>>>> asterisk-video mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>    http://lists.digium.com/mailman/listinfo/asterisk-video
>>>>
>>>>   
>>>>       
>>>     
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-video
>>
>>   
> 
>