[Asterisk-video] Problem with calling out to 3G handset via ISDN
Sergio Garcia
sergio.garcia at fontventa.com
Mon Aug 20 01:56:10 CDT 2007
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 --------------------
>
>Regards,
>
>Arnold P. Siboro (asiboro at maltech.jp)
>
>Technology is dominated by two types of people:
>Those who understand what they do not manage.
>Those who manage what they do not understand.
> -- Putt's Law
>
>
>_______________________________________________
>--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
>
More information about the asterisk-video
mailing list