[Asterisk-video] Problem with calling out to 3G handset via ISDN
Sergio Garcia
sergio.garcia at fontventa.com
Mon Aug 27 01:44:13 CDT 2007
I think that the best way is starting a bug in the bugtrack
http://sip.fontventa.com/trac/asterisk/log/
and attach the patch to it, I'll review it as soon as posssible and
commit it into the code.
Greetings and thank you
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, 27 Aug 2007 10:15:54 +0900
>
>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
>
>
>_______________________________________________
>--Bandwidth and Colocation Provided by http://www.api-digital.com--
>
>asterisk-video mailing list
>To UNSUBSCRIBE or update options visit:
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