[Asterisk-video] 3G-324M video stack which can work with asterisk
IvánF G
ctz.ivanf.bis at gmail.com
Fri May 8 10:33:34 CDT 2009
Hi all...
I think that I've advanced a little bit with my problem...
Certainly, the libpri-1.4.10 need to be patched for LLC support, but the
13055-libpri-1.4.4-llc-transmit-receive-patch-0.1.txt<http://bugs.digium.com/file_download.php?file_id=19452&type=bug>looks
to do the job... (the patch is applied perfectly)...
About Asterisk patch for *user information layer 1*, after applied it
manually (as I explain in the other post) and solved a couple of
complilation errors, seems to work too... (*CHANNEL(userinformationlayer1)=38")
*can be defined)
Nevertheless, outbound calls (neither inbound one, in fact...) work. The
cell phone receives the call but the dialplan doesn't executes anything at
all. Seems that the *user information layer1* field is not send, as you can
see here:
Asterisk 1.4 (outbound call OK):
[26G [2 at ive [42G [16Gexit [K originate local/3695563176 extension
803499621 at Context
ServerAst*CLI>
-- Executing [3695563176 at default:1]
h324m_call("Local/3695563176 at default-2852,2", "03695563176 at salientes_video")
in new stack
-- Executing [03695563176 at salientes_video:1]
Set("Local/03695563176 at salientes_video-b405,2",
"CHANNEL(transfercapability)=DIGITAL") in new stack
-- Executing [03695563176 at salientes_video:2]
NoOp("Local/03695563176 at salientes_video-b405,2", "transfer=DIGITAL") in new
stack
-- Executing [03695563176 at salientes_video:3]
Set("Local/03695563176 at salientes_video-b405,2",
"CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [03695563176 at salientes_video:4]
NoOp("Local/03695563176 at salientes_video-b405,2", "uil1=38") in new stack
-- Executing [03695563176 at salientes_video:5]
Dial("Local/03695563176 at salientes_video-b405,2", "Zap/g3/695563176") in new
stack
-- Making new call for cr 32770
-- digital call, setting user information layer 1 to 38 (0x26)
-- zap call: h324musellc=0, ast->userinformationlayer1=38
-- Requested transfer capability: 0x08 - DIGITAL
> [ 00 01 00 00 08 02 00 02 05 04 03 88 90 a6 18 03 a9 83 81 6c 02 00 c3 70
0a a1 36 39 35 35 36 33 31 37 36 a1 ]
> Informational frame:
> SAPI: 00 C/R: 0 EA: 0
> TEI: 000 EA: 1
> N(S): 000 0: 0
> N(R): 000 P: 0
> 32 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
> Protocol Discriminator: Q.931 (8) len=32
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: SETUP (5)
> [04 03 88 90 a6]
> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Unrestricted digital information (8)
> Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
> User information layer 1: H.223/H.245
Multimedia (38)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive
Dchan: 0
> ChanSel: Reserved
> Ext: 1 Coding: 0 Number Specified Channel Type: 3
> Ext: 1 Channel: 1 ]
Asterisk 1.6 (outbound call KO):
[Kserverast1*CLI> exit originate local/1695563176 extension 204
serverast1*CLI> -- Executing [1695563176 at default:1]
h324m_call("Local/1695563176 at default-1d04;2", "01695563176 at salientes_video")
in new stack
[May 8 14:33:55] DEBUG[21714]: app_h324m.c:1137 app_h324m_call: h324m_call
-- Executing [01695563176 at salientes_video:1]
Set("Local/01695563176 at salientes_video-3e49;2",
"CHANNEL(transfercapability)=DIGITAL") in new stack
-- Executing [01695563176 at salientes_video:2]
NoOp("Local/01695563176 at salientes_video-3e49;2", "transfer=DIGITAL") in new
stack
-- Executing [01695563176 at salientes_video:3]
Set("Local/01695563176 at salientes_video-3e49;2",
"CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [01695563176 at salientes_video:4]
NoOp("Local/01695563176 at salientes_video-3e49;2", "uil1=38") in new stack
-- Executing [01695563176 at salientes_video:5]
Dial("Local/01695563176 at salientes_video-3e49;2", "DAHDI/g1/695563176") in
new stack
[May 8 14:33:55] DEBUG[21715]: chan_dahdi.c:6302 dahdi_new: dahdi_new:
ps.curlaw=DAHDI_LAW_ALAW, setting deflaw to AST_FORMAT_ALAW
-- Making new call for cr 32773
-- digital call, setting user information layer 1 to 38 (0x26)
-- dahdi call: h324musellc=0, ast->userinformationlayer1=38
-- Requested transfer capability: 0x08 - DIGITAL
> [ 00 01 2a 3e 08 02 00 05 05 04 02 88 90 18 03 a9 83 81 6c 02 00 c3 70 0a
a1 36 39 35 35 36 33 31 37 36 a1 ]
> Informational frame:
> SAPI: 00 C/R: 0 EA: 0
> TEI: 000 EA: 1
> N(S): 021 0: 0
> N(R): 031 P: 0
> 31 bytes of data
Stopping T_203 timer
Starting T_200 timer
-- Restarting T200 timer
> Protocol Discriminator: Q.931 (8) len=31
> Call Ref: len= 2 (reference 5/0x5) (Originator)
> Message type: SETUP (5)
> [04 02 88 90]
> Bearer Capability (len= 4) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Unrestricted digital information (8)
> Ext: 1 Trans mode/rate: 64kbps, circuit-mode
(16)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive
Dchan: 0
> ChanSel: As indicated in following octets
> Ext: 1 Coding: 0 Number Specified Channel Type: 3
> Ext: 1 Channel: 1 ]
Any hint or suggestion?
Thanks and nice weekend...
2009/5/7 Dan Julius <dan.julius at gmail.com>
> Hi,
>
> Don't know about 1.6.x
>
> 1.4.24.1 works for me, unfortunately I'm not sure exactly which patches I
> had to apply.
> I could look into this if needed.
>
> Dan
>
>
> On Thu, May 7, 2009 at 5:43 PM, IvánF G <ctz.ivanf.bis at gmail.com> wrote:
>
>> Hi all...
>>
>> I've got a few questions related with this threat...
>>
>> I've got an asterisk-1.4.21.1 running on a debian machine with digium
>> hardware and fully operative for outbounds video calls (patched libpri,
>> patched asterisk, sip.fontventa support for amr, h324m, etc...)
>>
>> Now, I'm trying to get a new asterisk instalation on a diferent machine
>> (debian + Wildcard TE220). I'm using Asterisk 1.6.0.9, libpri-1.4.10,
>> dahdi-2.1.0.4, etc... and lastest code for the apps and support from
>> sip.fontventa.
>>
>> I can confirm that AMR and MP4 support work perfectly... libh324m and
>> app_h324m look operative too (after I solved a problem with a segmentation
>> fault problem related to app_h324m; search for '*asterisk startup
>> problems with latest app_h324m*')
>>
>> The question at this point is the needed patches of libpri and asterisk
>> for LLC and userinformationlayer support (needed in previous version at
>> less)... This patches are documented at:
>>
>> libpri: http://bugs.digium.com/view.php?id=13055
>> asterisk: http://bugs.digium.com/view.php?id=10217
>>
>> The last libpri patch is for version 1.4.4... I don't know if the
>> libpri-1.4.10 has this problem solved... Does anybody knows it? (I've
>> applied the oficial libpri-1.4.10-patch too...)
>>
>> About the asterisk link, the last version of the patch is really a mess an
>> is imposible to apply it with *patch *command (the result is
>> incoherent...)... I've tried to apply it manually, trying to compare it with
>> the last version of the 1.4 patch, but the compilation of asterisk crashes
>> (as I afraid...)
>>
>> Does anybody knows if there is a valid patch for the 1.6.X versions of
>> asterisk? Anybody has it working in this version? If the answer is not...
>> which is the last asterisk version that can be *officially *patched to
>> make the outbounds video calls work?
>>
>> As usual, thanks in advance and best regards...
>>
>> 2009/5/7 3g 2sip <3g2sip at gmail.com>
>>
>> Sergio,
>>> Thanks for your good news. Another question: can we do the video call
>>> test between two E1/ TE407P end points?
>>> Best Regards
>>> Mark
>>>
>>> On Thu, May 7, 2009 at 3:49 PM, Sergio Garcia Murillo <
>>> sergio.garcia at fontventa.com> wrote:
>>>
>>>> Hi Mark,
>>>>
>>>> Yes it shoudl work with asterisk 16.0.1 and next versions. If you find
>>>> any problem just let me know.
>>>>
>>>> Best regards
>>>> Sergio
>>>>
>>>> 3g 2sip escribió:
>>>>
>>>> Hi,Dan,
>>>> Thanks for your information, go through these information, found some
>>>> notes: *Note: Currently only Asterisk 1.4 is supported.*
>>>> Do you know the status of this project, Now we are using Asterisk 1.6,
>>>> can we use it? thanks.
>>>>
>>>> Mark
>>>>
>>>>
>>>> On Wed, May 6, 2009 at 2:15 PM, Dan Julius <dan.julius at gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> check out http://sip.fontventa.com
>>>>>
>>>>> Dan
>>>>>
>>>>>
>>>>> On Wed, May 6, 2009 at 6:07 AM, 3g 2sip <3g2sip at gmail.com> wrote:
>>>>>
>>>>>> Hello,everyone,
>>>>>> Does here has some source code and docs for 3G-324M video stack? or
>>>>>> any information related it, thanks.
>>>>>> Regards
>>>>>> Mark
>>>>>>
>>>>>> _______________________________________________
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>>>>>>
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>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
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>>>>>
>>>>
>>>> ------------------------------
>>>>
>>>>
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>>>>
>>>>
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>>>
>>>
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>>
>>
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>>
>
>
> _______________________________________________
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>
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