[Asterisk-video] Problem with calling out to 3G handset via ISDN
Sergio Garcia
sergio.garcia at fontventa.com
Thu Aug 23 01:47:51 CDT 2007
Hi Arnold
Be a bit pattient with the bug, in september I plan to start some intensive
testing to fix most of the bugs and features missing in h324m and app_mp4.
I'm in talks to be able to have access to some equipment for testing and
waiting for many people to come back from holidays.
Send me the patch, I'll review it and if you keep involved in the proyect
I will open an svn account for commiting. Also I have just setup a bugtrack
tool in the web so we can have a much better control of the issues.
BR
Sergio
---------- Original Message ----------------------------------
From: "Arnold P. Siboro" <asiboro at maltech.jp>
Reply-To: Development discussion of video media support in Asterisk<asterisk-video at lists.digium.com>
Date: Thu, 23 Aug 2007 09:18:13 +0900
>
>Sergio,
>
>I still could not make the call out to work on
>Asterisk+bristuff+libh324m. All patches known to this mailing list have
>been applied.
>
>BTW, is there any code review rule and who can commit to the repository?
>Why don't you use sourceforge for example so that bug report and patches
>can be better managed? I have a fix that I would like to be reviewed and
>patched into the source tree, who should I do it?
>
>Regards,
>
>Pada Mon, 20 Aug 2007 09:17:43 +0200
>si "Sergio Garcia" <sergio.garcia at fontventa.com> bilang:
>
>>
>> I don't know if the patches has been yet commitet to the main
>> asterisk code, try to take a look at it and apply the patch if not.
>>
>> BR.
>> Sergio
>>
>> ---------- Original Message ----------------------------------
>> From: "Arnold P. Siboro" <asiboro at maltech.jp>
>> Reply-To: Development discussion of video media support in Asterisk<asterisk-video at lists.digium.com>
>> Date: Mon, 20 Aug 2007 16:07:11 +0900
>>
>> >
>> >I think the patch from this email is already properly applied.
>> >http://lists.digium.com/pipermail/asterisk-video/2007-July/000858.html
>> >Klaus, any idea?
>> >
>> >Pada Mon, 20 Aug 2007 08:56:10 +0200
>> >si "Sergio Garcia" <sergio.garcia at fontventa.com> bilang:
>> >
>> >> Have you also applied the patch that Klaus sent in the same mail?
>> >>
>> >> BR
>> >> Sergio
>> >>
>> >>
>> >> ---------- Original Message ----------------------------------
>> >> From: "Arnold P. Siboro" <asiboro at maltech.jp>
>> >> Reply-To: Development discussion of video media support in Asterisk<asterisk-video at lists.digium.com>
>> >> Date: Mon, 20 Aug 2007 10:06:38 +0900
>> >>
>> >> >
>> >> >Hi all,
>> >> >
>> >> >I managed to build a system from libh324m that can receive calls from 3G
>> >> >handset via ISDN BRI. However I could not make a call out to 3G handset
>> >> >yet. I know some of you talked about how to make call out like this and
>> >> >I think I have followed them, but please tell me which I missed.
>> >> >
>> >> >Here is my extension.conf:
>> >> >[sipout]
>> >> >;exten => _X.,1,Dial(${GLOBAL(OUTGOING)}/${EXTEN})
>> >> >;exten => _X.,2,h324m_gw(start at videocall)
>> >> >;exten => _X.,2,h324m_call()
>> >> >;exten => _X.,3,Hangup()
>> >> >
>> >> >exten => _X.,1,Set(CHANNEL(transfercapability)=VIDEO)
>> >> >exten => _X.,n,NoOp(transfer=${CHANNEL(transfercapability)})
>> >> >exten => _X.,n,Set(CHANNEL(userinformationlayer1)=24)
>> >> >exten => _X.,n,NoOp(ul1=${CHANNEL(userinformationlayer1)})
>> >> >exten => _X.,n,Dial(${GLOBAL(OUTGOING)}/${EXTEN})
>> >> >exten => _X.,n,h324m_call()
>> >> >
>> >> >The SIP client tried to call 3G handset but failed with cause 96.
>> >> >Here is the log where error appeared:
>> >> >
>> >> >----------- LOG STARTED----------------------
>> >> >-- Executing [08017734983 at sipout:1] Set("SIP/172.31.31.98-092ef960",
>> >> >"CHANNEL(transfercapability)=VIDEO") in new stack
>> >> > -- Executing [08017734983 at sipout:2]
>> >> >NoOp("SIP/172.31.31.98-092ef960", "transfer=VIDEO") in new stack
>> >> > -- Executing [08017734983 at sipout:3] Set("SIP/172.31.31.98-092ef960",
>> >> >"CHANNEL(userinformationlayer1)=24") in new stack
>> >> > -- Executing [08017734983 at sipout:4]
>> >> >NoOp("SIP/172.31.31.98-092ef960", "ul1=24") in new stack
>> >> > -- Executing [08017734983 at sipout:5]
>> >> >Dial("SIP/172.31.31.98-092ef960", "Zap/1/08017734983") in new stack
>> >> >1 -- Making new call for cr 130
>> >> > -- digital call, setting user information layer 1 to 24 (0x18)
>> >> > -- Requested transfer capability: 0x18 - VIDEO
>> >> >1 -- Restarting T203 counter
>> >> >1 > Protocol Discriminator: Q.931 (8) len=34
>> >> >1 > Call Ref: len= 1 (reference 2/0x2) (Originator)
>> >> >1 > Message type: SETUP (5)
>> >> >1 > [04 03 88 90 98]
>> >> >1 > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
>> >> >capability: Unrestricted digital information (8)
>> >> >1 > Ext: 1 Trans mode/rate: 64kbps,
>> >> >circuit-mode (16)
>> >> >1 > Ext: 1 User information layer 1:
>> >> >Unknown (24)
>> >> >1 > [18 01 81]
>> >> >1 > Channel ID (len= 3) [ Ext: 1 IntID: Implicit Other Spare: 0
>> >> >Preferred Dchan: 0
>> >> >1 > ChanSel: B1 channel
>> >> >1 ]
>> >> >1 > [6c 06 41 80 31 30 30 39]
>> >> >1 > Calling Number (len= 8) [ Ext: 0 TON: Subscriber Number (4) NPI:
>> >> >ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>> >> >1 > Presentation: Presentation permitted, user
>> >> >number not screened (0) '1009' ]
>> >> >1 > [70 0b a1 38 30 31 37 37 33 34 39 38 33]
>> >> >1 > Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI:
>> >> >ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8017734983' ]
>> >> >1 > [a1]
>> >> >1 > Sending Complete (len= 1)
>> >> >1 q931.c:3638 q931_setup: call 130 on channel 1 enters state 1 (Call
>> >> >Initiated)
>> >> > -- Called 1/08017734983
>> >> >1 -- Restarting T203 counter
>> >> >1 -- Restarting T203 counter
>> >> >1 < Protocol Discriminator: Q.931 (8) len=9
>> >> >1 < Call Ref: len= 1 (reference 130/0x82) (Terminator)
>> >> >1 < Message type: RELEASE COMPLETE (90)
>> >> >1 < [08 03 82 e0 04]
>> >> >1 < Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
>> >> >Location: Public network serving the local user (2)
>> >> >1 < Ext: 1 Cause: Mandatory information element is
>> >> >missing (96), class = Protocol Error (e.g. unknown message) (6) ]
>> >> >1 < Cause data 1: 04 (4)
>> >> >1 -- Processing IE 8 (cs0, Cause)
>> >> >1 q931.c:4483 q931_receive: call 130 on channel 1 enters state 0 (Null)
>> >> >1 -- Restarting T203 counter
>> >> > -- Channel 0/1, span 1 got hangup, cause 96
>> >> >1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
>> >> >1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
>> >> > -- Hungup 'Zap/1-1'
>> >> > == Everyone is busy/congested at this time (1:0/0/1)
>> >> > -- Executing [08017734983 at sipout:6]
>> >> >h324m_call("SIP/172.31.31.98-092ef960", "") in new stack
>> >> > == Spawn extension (sipout, 08017734983, 6) exited non-zero on
>> >> >'SIP/172.31.31.98-092ef960'
>> >> >
>> >> >
>> >> >--------------------- Original Message Ends --------------------
>> >
>
>
>Arnold P. Siboro (asiboro at maltech.jp)
>
>"A tidy desk is a sign of untidy mind"
> --The Sputtering R&D Machine
> The Innovative Enterprise Aug 2002 - Harvard Business Review
>
>
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