[Asterisk-video] Patch 0010217
Valerio Puglia
valerio at oscorp.sm
Tue Mar 18 06:42:37 CDT 2008
it's work!tnx
Klaus Darilion ha scritto:
> 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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>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-video
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>>>
>>>>> asterisk-video mailing list
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>>>>> http://lists.digium.com/mailman/listinfo/asterisk-video
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>> _______________________________________________
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>>>
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>>>
>>>
>>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
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>>
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>>
>
> _______________________________________________
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>
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>
>
--
Best regards,
Valerio Puglia
OScorp S.P.A.
NETWORK Adm
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