[Asterisk-video] Problem with calling out to 3G handset via ISDN
Arnold P. Siboro
asiboro at maltech.jp
Tue Aug 28 02:22:11 CDT 2007
I used Nokia 6680 and Samsung 804SS. I just sent you the logs.
Regards,
Pada Tue, 28 Aug 2007 07:31:54 +0200
si "Sergio Garcia Murillo" <sergio.garcia at fontventa.com> bilang:
> Which handset are u using?
> Could you run asterisk with -fdddvvv and paste the h245 negotiation?
> Also send me the h223 log privately to take a look
>
> BR
> Sergio
>
> ----- Original Message -----
> From: "Arnold P. Siboro" <asiboro at maltech.jp>
> To: <asterisk-video at lists.digium.com>
> Sent: Tuesday, August 28, 2007 1:18 AM
> Subject: Re: [Asterisk-video] Problem with calling out to 3G handset via
> ISDN
>
>
> >
> > Regarding the ISDN outgoing video call problem, we have noticed that
> > Q.931 signaling might be different in Japan.
> > We observed that incoming video call is coded with user information
> > layer 1 of 24 (instead of 38 like Klaus mentioned here
> http://lists.digium.com/pipermail/asterisk-video/2007-July/000858.html), and
> with
> > capability set on the low layer. We tried to fix the chan_zap.c to set
> > the bearer capability on the low layer, outgoing call was made to the 3G
> > handset but the handset kept waiting for image to come, and ISDN
> > debugging shows that there is problem with H.245.
> > Any idea?
> >
> >
> >
> > 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)
> >
> > "Imagination is more important than knowledge."
> > --Albert Einstein.
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-video mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-video
> >
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
Arnold P. Siboro (asiboro at maltech.jp)
"A lie can travel half-way around the world while truth puts on its shoes."
-- Mark Twain (1835 - 1910)
More information about the asterisk-video
mailing list