Posted: Thu Mar 13, 2008 5:32 pm Post subject: [Asterisk-video] Patch 0010217
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?
--
Best regards,
Valerio Puglia
OScorp S.P.A.
NETWORK Adm
_______________________________________________
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Posted: Thu Mar 13, 2008 9:54 pm Post subject: [Asterisk-video] Patch 0010217
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:
Quote:
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--
Posted: Fri Mar 14, 2008 9:44 am Post subject: [Asterisk-video] Patch 0010217
hi Klaus
i remove AST_FORMAT_ULAW and it works
but when bridge 2 mobile thelephone or call from sipphone to meobile
phone the video doesn't start.....
Klaus Darilion ha scritto:
Quote:
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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Posted: Fri Mar 14, 2008 10:16 am Post subject: [Asterisk-video] Patch 0010217
AST_FRAME_DIGITAL anyone?? jejeje
By the way Klaus, is any change on 1.6 that we could use on that issue?
Best regards
Sergio
----- Original Message -----
From: Valerio Puglia [mailto:valerio@oscorp.sm]
To: asterisk-video@lists.digium.com
Sent: Fri, 14 Mar 2008 10:36:59 +0100
Subject: Re: [Asterisk-video] Patch 0010217
hi Klaus
i remove AST_FORMAT_ULAW and it works
but when bridge 2 mobile thelephone or call from sipphone to meobile
phone the video doesn't start.....
Klaus Darilion ha scritto:
Quote:
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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Posted: Fri Mar 14, 2008 12:22 pm Post subject: [Asterisk-video] Patch 0010217
Valerio Puglia schrieb:
Quote:
hi Klaus
i remove AST_FORMAT_ULAW and it works
what does work?
Quote:
but when bridge 2 mobile thelephone or call from sipphone to meobile
phone the video doesn't start.....
I do not understand - above your write that it works now?
klaus
Quote:
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
>
>
_______________________________________________
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Posted: Fri Mar 14, 2008 12:35 pm Post subject: [Asterisk-video] Patch 0010217
Sergio Garcia Murillo schrieb:
Quote:
AST_FRAME_DIGITAL anyone?? jejeje
By the way Klaus, is any change on 1.6 that we could use on that issue?
No. I still would favor AST_FORMAT_DIGITAL for "digital" voice calls but
that would require to patch every ISDN channel driver.
I wonder if outgoing 3G calls work for anyone without setting ALAW
manually in app_h324m.c.
Last time I tried to find the cause of the problem and it is:
- h324m_call requests a local channel in format ALAW|ULAW
- the local channel traverses Asterisk's preferred codecs list and
decides to use ULAW (Asterisk preferres ULAW)
- when the zap channel is requested, chan zap detects it is a "digital"
call and thus does not set a certain codec, but uses the "deflaw"
setting of the zaptel kernel module - e.g. using PRI+E1 or bristuff the
deflaw is ALAW.
- now Asterisk has a zap channel with ALAW and a Local channel with ULAW
and starts transcoding :-(
I think most app_h324m users reside in Europe, thus have a deflaw of
ALAW. Wouldn't it be better to make ALAW the default in h324m_call, or
make a command line option to set the "tunnel-codec"?
regards
klaus
Quote:
Best regards
Sergio
----- Original Message -----
From: Valerio Puglia [mailto:valerio@oscorp.sm]
To: asterisk-video@lists.digium.com
Sent: Fri, 14 Mar 2008 10:36:59 +0100
Subject: Re: [Asterisk-video] Patch 0010217
hi Klaus
i remove AST_FORMAT_ULAW and it works
but when bridge 2 mobile thelephone or call from sipphone to meobile
phone the video doesn't start.....
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
>
>
Posted: Fri Mar 14, 2008 1:59 pm Post subject: [Asterisk-video] Patch 0010217
Hi Klaus
Quote:
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
Quote:
> 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
Spawn extension (call2, 666, 1) exited non-zero on
'Local/666@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@from-pstn:1] h324m_call("Zap/116-1",
"666@video-out2") in new stack
-- Executing [666@video-out2:1] Set("Local/666@video-out2-45df,2",
"CHANNEL(transfercapability)=VIDEO") in new stack
-- Executing [666@video-out2:2] NoOp("Local/666@video-out2-45df,2",
"transfer=VIDEO") in new stack
-- Executing [666@video-out2:3] Set("Local/666@video-out2-45df,2",
"CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [666@video-out2:4] NoOp("Local/666@video-out2-45df,2",
"ul1=38") in new stack
-- Executing [666@video-out2:5] Goto("Local/666@video-out2-45df,2",
"call2|666|1") in new stack
-- Goto (call2,666,1)
-- Executing [666@call2:1] Dial("Local/666@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@video-out2-45df,2
== Spawn extension (call2, 666, 1) exited non-zero on
'Local/666@video-out2-45df,2'
-- Channel 0/23, span 4 got hangup request, cause 16
-- Hungup 'Zap/94-1'
-- Hungup 'Zap/116-1
Quote:
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--
Posted: Mon Mar 17, 2008 8:52 am Post subject: [Asterisk-video] Patch 0010217
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.
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.
> 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
Spawn extension (call2, 666, 1) exited non-zero on
'Local/666@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@from-pstn:1] h324m_call("Zap/116-1",
"666@video-out2") in new stack
-- Executing [666@video-out2:1] Set("Local/666@video-out2-45df,2",
"CHANNEL(transfercapability)=VIDEO") in new stack
-- Executing [666@video-out2:2] NoOp("Local/666@video-out2-45df,2",
"transfer=VIDEO") in new stack
-- Executing [666@video-out2:3] Set("Local/666@video-out2-45df,2",
"CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [666@video-out2:4] NoOp("Local/666@video-out2-45df,2",
"ul1=38") in new stack
-- Executing [666@video-out2:5] Goto("Local/666@video-out2-45df,2",
"call2|666|1") in new stack
-- Goto (call2,666,1)
-- Executing [666@call2:1] Dial("Local/666@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@video-out2-45df,2
== Spawn extension (call2, 666, 1) exited non-zero on
'Local/666@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
>
>
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Posted: Mon Mar 17, 2008 10:25 am Post subject: [Asterisk-video] Patch 0010217
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:
Quote:
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.
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.
> 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@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@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@from-pstn:1] h324m_call("Zap/116-1",
> "666@video-out2") in new stack
> -- Executing [666@video-out2:1] Set("Local/666@video-out2-45df,2",
> "CHANNEL(transfercapability)=VIDEO") in new stack
> -- Executing [666@video-out2:2] NoOp("Local/666@video-out2-45df,2",
> "transfer=VIDEO") in new stack
> -- Executing [666@video-out2:3] Set("Local/666@video-out2-45df,2",
> "CHANNEL(userinformationlayer1)=38") in new stack
> -- Executing [666@video-out2:4] NoOp("Local/666@video-out2-45df,2",
> "ul1=38") in new stack
> -- Executing [666@video-out2:5] Goto("Local/666@video-out2-45df,2",
> "call2|666|1") in new stack
> -- Goto (call2,666,1)
> -- Executing [666@call2:1] Dial("Local/666@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@video-out2-45df,2
> == Spawn extension (call2, 666, 1) exited non-zero on
> 'Local/666@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
>>
>>
>>
>
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
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@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@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@from-pstn:1] h324m_call("Zap/116-1",
>> "666@video-out2") in new stack
>> -- Executing [666@video-out2:1]
>> Set("Local/666@video-out2-45df,2",
>> "CHANNEL(transfercapability)=VIDEO") in new stack
>> -- Executing [666@video-out2:2]
>> NoOp("Local/666@video-out2-45df,2", "transfer=VIDEO") in new stack
>> -- Executing [666@video-out2:3]
>> Set("Local/666@video-out2-45df,2",
>> "CHANNEL(userinformationlayer1)=38") in new stack
>> -- Executing [666@video-out2:4]
>> NoOp("Local/666@video-out2-45df,2", "ul1=38") in new stack
>> -- Executing [666@video-out2:5]
>> Goto("Local/666@video-out2-45df,2", "call2|666|1") in new stack
>> -- Goto (call2,666,1)
>> -- Executing [666@call2:1] Dial("Local/666@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@video-out2-45df,2
>> == Spawn extension (call2, 666, 1) exited non-zero on
>> 'Local/666@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
>>>
>>>
>>
>
> _______________________________________________
> --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
>
>
> 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@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@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@from-pstn:1] h324m_call("Zap/116-1",
>>> "666@video-out2") in new stack
>>> -- Executing [666@video-out2:1]
>>> Set("Local/666@video-out2-45df,2",
>>> "CHANNEL(transfercapability)=VIDEO") in new stack
>>> -- Executing [666@video-out2:2]
>>> NoOp("Local/666@video-out2-45df,2", "transfer=VIDEO") in new stack
>>> -- Executing [666@video-out2:3]
>>> Set("Local/666@video-out2-45df,2",
>>> "CHANNEL(userinformationlayer1)=38") in new stack
>>> -- Executing [666@video-out2:4]
>>> NoOp("Local/666@video-out2-45df,2", "ul1=38") in new stack
>>> -- Executing [666@video-out2:5]
>>> Goto("Local/666@video-out2-45df,2", "call2|666|1") in new stack
>>> -- Goto (call2,666,1)
>>> -- Executing [666@call2:1] Dial("Local/666@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@video-out2-45df,2
>>> == Spawn extension (call2, 666, 1) exited non-zero on
>>> 'Local/666@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
>>>>
>>>>
>>>>
>>>
>>>
>> _______________________________________________
>> --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
>
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