Posted: Sun May 24, 2009 1:18 pm Post subject: [Asterisk-video] 3G-324M video stack which can work with ast
Hi all,
I keep facing the problem descripted in my last mail (missing 5º octet in Bearer Capability when I try to do an outbound call) and I have problems too with incoming video calls; in this case the user information layer looks to be correctly received but Asterisk goes freeze at h324m_gw_answer... (see trace below)
Anyone has this scenary working (Asterisk 1.6.X + h324m stack from sip.fontventa) for incoming/outcoming video calls? Am I trying to get working anything not supported yet?
< Informational frame:
< SAPI: 00 C/R: 1 EA: 0
< TEI: 000 EA: 1
< N(S): 016 0: 0
< N(R): 013 P: 0
< 46 bytes of data
Handling message for SAPI/TEI=0/0
-- ACKing all packets from 12 to (but not including) 13
-- Since there was nothing left, stopping T200 counter
-- Stopping T203 counter since we got an ACK
-- Nothing left, starting T203 counter
[Kserverast1*CLI> < Protocol Discriminator: Q.931 (8) len=46
< Call Ref: len= 2 (reference 8704/0x2200) (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 a1 83 85]
[Kserverast1*CLI> < Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Preferred Dchan: 0
< ChanSel: As indicated in following octets
< Ext: 1 Coding: 0 Number Specified Channel Type: 3
< Ext: 1 Channel: 5 ]
< [6c 0b 21 83 36 39 35 35 36 33 31 37 36]
< Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
< Presentation: Presentation allowed of network provided number (3) '695563176' ]
< [70 0a a1 39 31 31 35 34 37 31 39 30]
< Called Number (len=12) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '911547190' ]
< [a1]
< Sending Complete (len= 1)
< [7c 03 88 90 a6]
< Low-layer Compatability (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)
-- Making new call for cr 8704
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
-- Processing IE 161 (cs0, Sending Complete)
-- Processing IE 124 (cs0, Low-layer Compatibility)
receive_low_layer_compatibility
LLC User layer 1: H.223/H.245 Multimedia (38)
q931.c:3751 q931_receive: call 8704 on channel 5 enters state 6 (Call Present)
Sending Receiver Ready (17)
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: 5 ]
[Kserverast1*CLI> -- Accepting call from '695563176' to '911547190' on channel 0/5, span 1
-- Executing [911547190@entrantes-pra1:1] GotoIf("DAHDI/5-1", "0?voz") in new stack
-- Executing [911547190@entrantes-pra1:2] GotoIf("DAHDI/5-1", "0?voz") in new stack
-- Executing [911547190@entrantes-pra1:3] h324m_gw("DAHDI/5-1", "204@default") in new stack
-- Executing [204@default:1] h324m_gw_answer("Local/204@default-0a31;2", "") in new stack
[Kserverast1*CLI> q931.c:3133 q931_connect: call 8704 on channel 5 enters state 8 (Connect Request)
[08 02 81 90]
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1)
Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]
-- Hungup 'DAHDI/5-1'
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@ive [42G [16Gexit [K originate local/3695563176 extension 803499621@Context
ServerAst*CLI>
-- Executing [3695563176@default:1] h324m_call("Local/3695563176@default-2852,2", "03695563176@salientes_video") in new stack
-- Executing [03695563176@salientes_video:1] Set("Local/03695563176@salientes_video-b405,2", "CHANNEL(transfercapability)=DIGITAL") in new stack
-- Executing [03695563176@salientes_video:2] NoOp("Local/03695563176@salientes_video-b405,2", "transfer=DIGITAL") in new stack
-- Executing [03695563176@salientes_video:3] Set("Local/03695563176@salientes_video-b405,2", "CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [03695563176@salientes_video:4] NoOp("Local/03695563176@salientes_video-b405,2", "uil1=38") in new stack
-- Executing [03695563176@salientes_video:5] Dial("Local/03695563176@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
[Kserverast1*CLI> exit originate local/1695563176 extension 204
serverast1*CLI> -- Executing [1695563176@default:1] h324m_call("Local/1695563176@default-1d04;2", "01695563176@salientes_video") in new stack
[May 8 14:33:55] DEBUG[21714]: app_h324m.c:1137 app_h324m_call: h324m_call
-- Executing [01695563176@salientes_video:1] Set("Local/01695563176@salientes_video-3e49;2", "CHANNEL(transfercapability)=DIGITAL") in new stack
-- Executing [01695563176@salientes_video:2] NoOp("Local/01695563176@salientes_video-3e49;2", "transfer=DIGITAL") in new stack
-- Executing [01695563176@salientes_video:3] Set("Local/01695563176@salientes_video-3e49;2", "CHANNEL(userinformationlayer1)=38") in new stack
-- Executing [01695563176@salientes_video:4] NoOp("Local/01695563176@salientes_video-3e49;2", "uil1=38") in new stack
-- Executing [01695563176@salientes_video:5] Dial("Local/01695563176@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
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@gmail.com (ctz.ivanf.bis@gmail.com)> wrote:
Quote:
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:
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?
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@fontventa.com (sergio.garcia@fontventa.com)> wrote:
Quote:
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:
Quote:
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@gmail.com (dan.julius@gmail.com)> wrote:
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