[Asterisk-video] Problem with calling out to 3G handset via ISDN
Arnold P. Siboro
asiboro at maltech.jp
Wed Aug 22 19:18:13 CDT 2007
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
More information about the asterisk-video
mailing list