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 => _X.,2,h324m_gw(start 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 => _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 [08017734983sipout:1]
Set("SIP/172.31.31.98-092ef960",
"CHANNEL(transfercapability)=VIDEO") in new stack
-- Executing [08017734983sipout:2]
NoOp("SIP/172.31.31.98-092ef960",
"transfer=VIDEO") in new stack
-- Executing [08017734983sipout:3]
Set("SIP/172.31.31.98-092ef960",
"CHANNEL(userinformationlayer1)=24") in new stack
-- Executing [08017734983sipout:4]
NoOp("SIP/172.31.31.98-092ef960",
"ul1=24") in new stack
-- Executing [08017734983sipout: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 [08017734983sipout: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 (asiboromaltech.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.c
om--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
|
