<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 TRANSITIONAL//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; CHARSET=UTF-8">
<META NAME="GENERATOR" CONTENT="GtkHTML/3.24.1.1">
</HEAD>
<BODY>
<BR>
Thanks Borja,<BR>
<BR>
This example worked like a charm!<BR>
<BR>
Joost<BR>
<BR>
<BR>
On Wed, 2008-11-19 at 00:40 +0100, Borja SIXTO wrote:
<BLOCKQUOTE TYPE=CITE>
<PRE>
Hi Joost,
Example of .call
Set: CHANNEL(transfercapability)=VIDEO
Channel: ZAP/g1/0033xxxxxxxx
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Extension: dialervideo
Priority: 1
Example of extension :
exten => dialervideo,1,Answer()
exten => dialervideo,n,h324m_gw(dialervideo_gw@default)
exten => dialervideo_gw,1,h324m_gw_answer()
exten => dialervideo_gw,n,mp4play(/root/dialervideo/message.3gp)
;exten => dialervideo_gw,n,vxml(dialervideo)
Regards,
Borja
<A HREF="mailto:borja.sixto@i6net.com">borja.sixto@i6net.com</A> a écrit :
> Hi,
>
> Do you use the h324m_call application ?
> Can you copy past the .call file and the extension section (you should jump to a
> extension service).
>
> Regards,
>
>
> Borja
>
> Selon Joost Kuif <<A HREF="mailto:joost.kuif@mobillion.nl">joost.kuif@mobillion.nl</A>>:
>
>
>> Hello people,
>>
>> I am trying to make outbound videocalls with asterisk 1.4.22 and dahdi.
>> For triggering the call i use the asterisk spool directory.
>> I am at the point where asterisk is able to dial out to the mobile, and
>> the h324m_loopback is working ok. audio and video are working ok with
>> the loopback function. See details below.
>>
>> When i change the h324m_loopback to the mp4play application and redial
>> again the mp4play function doesn't play audio or video.
>>
>> One thing that gets my attention: why seems the transfercapability of
>> the channel in both cases SPEECH? the CLI logging says VIDEO is
>> requested for the transfercapability. Am i missing a patch? I did the
>> amr patch, the userlayer1 patch. The dahdi driver 2.0.0 and libpri 1.4.7
>> seems to already have the changes that last year had to be patched
>> manually.
>>
>> Is it possible at all to initiate a video call and use mp4play through
>> the spool mechanism of asterisk? If not, what mechanism do you prefer if
>> you want to dial out automated (without dialing out initiated via a sip
>> client or whatsoever)? AMI interface maybe?
>>
>>
>> Regards and thanks for your help,
>>
>> Joost Kuif
>>
>> ---------------------------
>>
>> Spool file:
>>
>> root@umts-test:/var/spool/asterisk/tmp# cat test.call
>> Channel: DAHDI/1/00316********
>> Callerid: <318******>
>> WaitTime: 30
>> MaxRetries: 0
>> RetryTime: 30
>> Set: CHANNEL(transfercapability)=VIDEO
>> Set: CHANNEL(userinformationlayer1)=38
>> Context: xcon
>> Extension: 665
>> Priority: 1
>>
>> Dialplan:
>> [xcon]
>> exten => 665,1,h324m_loopback()
>>
>>
>>
>> umts-test*CLI>
>> -- Attempting call on DAHDI/1/00316******** for 665@xcon:1 (Retry 1)
>> -- digital call, setting user information layer 1 to 38 (0x26)
>> -- dahdi call: h324musellc=0, ast->userinformationlayer1=38
>> -- Requested transfer capability: 0x18 - VIDEO
>> > Channel DAHDI/1-1 was answered.
>> -- Executing [665@xcon:1] h324m_loopback("DAHDI/1-1", "") in new
>> stack
>> umts-test*CLI> core show channel DAHDI/1-1
>> -- General --
>> Name: DAHDI/1-1
>> Type: DAHDI
>> UniqueID: 1227003199.1
>> Caller ID: 318631904
>> Caller ID Name: (N/A)
>> DNID Digits: (N/A)
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x48 (alaw|slin)
>> WriteFormat: 0x8 (alaw)
>> ReadFormat: 0x8 (alaw)
>> WriteTranscode: No
>> ReadTranscode: No
>> 1st File Descriptor: 21
>> Frames in: 1099
>> Frames out: 890
>> Time to Hangup: 0
>> Elapsed Time: 0h0m22s
>> Direct Bridge: <none>
>> Indirect Bridge: <none>
>> -- PBX --
>> Context: xcon
>> Extension: 665
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: h324m_loopback
>> Data: (Empty)
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> TRANSFERCAPABILITY=SPEECH
>>
>> CDR Variables:
>> level 1: dst=s
>> level 1: dcontext=incoming
>> level 1: channel=DAHDI/1-1
>> level 1: lastapp=h324m_loopback
>> level 1: start=2008-11-18 11:13:19
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=NO ANSWER
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1227003199.1
>>
>> -- Channel 0/1, span 1 got hangup request, cause 16
>> -- Hungup 'DAHDI/1-1'
>> [Nov 18 11:14:20] NOTICE[13227]: pbx_spool.c:365 attempt_thread: Call
>> completed to DAHDI/1/0031646160590
>> umts-test*CLI>
>>
>>
>>
>> When i change the h324m_loopback to the mp4play application and redial
>> again the mp4play function doesn't play audio or video:
>>
>> umts-test*CLI> core show channel DAHDI/1-1
>> -- General --
>> Name: DAHDI/1-1
>> Type: DAHDI
>> UniqueID: 1227016348.0
>> Caller ID: 318631904
>> Caller ID Name: (N/A)
>> DNID Digits: (N/A)
>> State: Up (6)
>> Rings: 0
>> NativeFormats: 0x48 (alaw|slin)
>> WriteFormat: 0x2000 (amr)
>> ReadFormat: 0x8 (alaw)
>> WriteTranscode: Yes
>> ReadTranscode: No
>> 1st File Descriptor: 21
>> Frames in: 1670
>> Frames out: 414
>> Time to Hangup: 0
>> Elapsed Time: 0h0m33s
>> Direct Bridge: <none>
>> Indirect Bridge: <none>
>> -- PBX --
>> Context: xcon
>> Extension: 665
>> Priority: 1
>> Call Group: 0
>> Pickup Group: 0
>> Application: mp4play
>> Data: /var/lib/asterisk/movies/umts_uitbellen/intro.3gp
>> Blocking in: ast_waitfor_nandfds
>> Variables:
>> TRANSFERCAPABILITY=SPEECH
>>
>> CDR Variables:
>> level 1: dst=s
>> level 1: dcontext=incoming
>> level 1: channel=DAHDI/1-1
>> level 1: lastapp=mp4play
>> level 1: lastdata=/var/lib/asterisk/movies/umts_uitbellen/intro.3gp
>> level 1: start=2008-11-18 14:52:28
>> level 1: duration=0
>> level 1: billsec=0
>> level 1: disposition=NO ANSWER
>> level 1: amaflags=DOCUMENTATION
>> level 1: uniqueid=1227016348.0
>>
>> umts-test*CLI>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by <A HREF="http://www.api-digital.com">http://www.api-digital.com</A>--
>>
>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>> <A HREF="http://lists.digium.com/mailman/listinfo/asterisk-video">http://lists.digium.com/mailman/listinfo/asterisk-video</A>
>>
>>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by <A HREF="http://www.api-digital.com">http://www.api-digital.com</A>--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> <A HREF="http://lists.digium.com/mailman/listinfo/asterisk-video">http://lists.digium.com/mailman/listinfo/asterisk-video</A>
>
</PRE>
</BLOCKQUOTE>
</BODY>
</HTML>