[Asterisk-video] Patch 0010217
Klaus Darilion
klaus.mailinglists at pernau.at
Mon Mar 17 07:55:54 CDT 2008
sorry, there was a type, use h324m_gw the first time:
[from-pstn]
exten => _X.,1,h324m_gw(${EXTEN}@h324m-decoded)
^^^^^^
[h324m-decoded]
exten => _X.,1,h324m_call(${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)
Valerio Puglia schrieb:
> 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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