[Asterisk-video] Problem with calling out to 3G handset via ISDN
Arnold P. Siboro
asiboro at maltech.jp
Sun Aug 26 20:15:54 CDT 2007
Hi Sergio,
OK. I am not being impatient, just thinking that the changes I made
should be reviewed and committed back to the repository when considered
useful.
I'll send the patch.
Regards,
Pada Thu, 23 Aug 2007 08:47:51 +0200
si "Sergio Garcia" <sergio.garcia at fontventa.com> bilang:
> 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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>
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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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