[Asterisk-video] [spam] Outbound Call / Callout Support

Fri Nov 2 11:58:34 CDT 2007

Hi Sergio,

hmm, after some research I got what you meant, sorry. Now I am using
SIPp.

But I have encountered an hangup issue trying to make outbound video
calls to 3G handsets (have a look at the logs below) . 

We integrated the patches of Klaus according to the Q.931 issues without
effect. Any further ideas?

Best Regards,

Cihan

### extensions.conf

[test]

exten => 665,1,h324m_call(666 at test)

exten => 666,1,Set(CHANNEL(transfercapability)=VIDEO)

exten => 666,n,NoOp(transfer=${CHANNEL(transfercapability)})

exten => 666,n,Set(CHANNEL(userinformationlayer1)=38)

exten => 666,n,NoOp(ul1=${CHANNEL(userinformationlayer1)})

exten => 666,n,Dial,Zap/g1/162...

### Log without pri debug span 1

    -- Executing [665 at test:1] h324m_call("SIP/sippuas-0823c430",
"666 at test") in new stack

    -- Executing [666 at test:1] Set("Local/666 at test-03fc,2",
"CHANNEL(transfercapability)=VIDEO") in new stack

    -- Executing [666 at test:2] NoOp("Local/666 at test-03fc,2",
"transfer=VIDEO") in new stack

    -- Executing [666 at test:3] Set("Local/666 at test-03fc,2",
"CHANNEL(userinformationlayer1)=38") in new stack

    -- Executing [666 at test:4] NoOp("Local/666 at test-03fc,2", "ul1=38") in
new stack

    -- Executing [666 at test:5] Dial("Local/666 at test-03fc,2",
"Zap/g1/162...") in new stack

    -- digital call, setting user information layer 1 to 38 (0x26)

    -- Requested transfer capability: 0x18 - VIDEO

    -- Called g1/162...

    -- Channel 0/1, span 1 got hangup request, cause 38

    -- Zap/1-1 is circuit-busy

    -- Hungup 'Zap/1-1'

  == Everyone is busy/congested at this time (1:0/1/0)

  == Auto fallthrough, channel 'Local/666 at test-03fc,2' status is
'CONGESTION'

  == Spawn extension (test, 665, 1) exited non-zero on
'SIP/sippuas-0823c430'

### Log with pri debug span 1

    -- Executing [665 at test:1] h324m_call("SIP/sippuas-0823c430",
"666 at test") in new stack

    -- Executing [666 at test:1] Set("Local/666 at test-01c4,2",
"CHANNEL(transfercapability)=VIDEO") in new stack

    -- Executing [666 at test:2] NoOp("Local/666 at test-01c4,2",
"transfer=VIDEO") in new stack

    -- Executing [666 at test:3] Set("Local/666 at test-01c4,2",
"CHANNEL(userinformationlayer1)=38") in new stack

    -- Executing [666 at test:4] NoOp("Local/666 at test-01c4,2", "ul1=38") in
new stack

    -- Executing [666 at test:5] Dial("Local/666 at test-01c4,2",
"Zap/g1/162...") in new stack

-- Making new call for cr 32796

    -- digital call, setting user information layer 1 to 38 (0x26)

    -- Requested transfer capability: 0x18 - VIDEO

> Protocol Discriminator: Q.931 (8)  len=35

> Call Ref: len= 2 (reference 28/0x1C) (Originator)

> Message type: SETUP (5)

> [04 02 88 90]

> Bearer Capability (len= 4) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Unrestricted digital information (8)

>                              Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)

>                              Ext: 0  User information layer 1: Unknown
(24)

> [18 03 a9 83 81]

> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
Exclusive  Dchan: 0

>                        ChanSel: Reserved

>                       Ext: 1  Coding: 0  Number Specified  Channel
Type: 3

>                       Ext: 1  Channel: 1 ]

> [6c 06 21 80 73 69 70 70]

> Calling Number (len= 8) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)

>                           Presentation: Presentation permitted, user
number not screened (0)  'sipp' ]

> [70 0b a1 31 36 32 32 31 31 32 37 38 30]

> Called Number (len=13) [ Ext: 1  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '162...' ]

q931.c:2879 q931_setup: call 32796 on channel 1 enters state 1 (Call
Initiated)

    -- Called g1/162...

< Protocol Discriminator: Q.931 (8)  len=14

< Call Ref: len= 2 (reference 28/0x1C) (Terminator)

< Message type: STATUS (125)

< [08 04 82 e4 98 6c]

< Cause (len= 6) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Public network serving the local user (2)

<                  Ext: 1  Cause: Invalid information element contents
(100), class = Protocol Error (e.g. unknown message) (6) ]

<              Cause data 1: 98 (152)

<              Cause data 2: 6c (108)

< [14 01 01]>

< Call State (len= 3) [ Ext: 0  Coding: CCITT (ITU) standard (0)  Call
state: Call Initiated (1)

-- Processing IE 8 (cs0, Cause)

-- Processing IE 20 (cs0, Call State)

< Protocol Discriminator: Q.931 (8)  len=10

< Call Ref: len= 2 (reference 28/0x1C) (Terminator)

< Message type: SETUP ACKNOWLEDGE (13)

< [18 03 a9 83 81]

< Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
Exclusive  Dchan: 0

<                        ChanSel: Reserved

<                       Ext: 1  Coding: 0  Number Specified  Channel
Type: 3

<                       Ext: 1  Channel: 1 ]

-- Processing IE 24 (cs0, Channel Identification)

q931.c:3620 q931_receive: call 32796 on channel 1 enters state 2
(Overlap sending)

< Protocol Discriminator: Q.931 (8)  len=9

< Call Ref: len= 2 (reference 28/0x1C) (Terminator)

< Message type: DISCONNECT (69)

< [08 02 8a a6]

< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Network beyond the interworking point (10)

<                  Ext: 1  Cause: Network out of order (38), class =
Network Congestion (resource unavailable) (2) ]

-- Processing IE 8 (cs0, Cause)

q931.c:3561 q931_receive: call 32796 on channel 1 enters state 12
(Disconnect Indication)

    -- Channel 0/1, span 1 got hangup request, cause 38

    -- Zap/1-1 is circuit-busy

NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
peerstate Disconnect Request

q931.c:2715 q931_release: call 32796 on channel 1 enters state 19
(Release Request)

> Protocol Discriminator: Q.931 (8)  len=9

> Call Ref: len= 2 (reference 28/0x1C) (Originator)

> Message type: RELEASE (77)

> [08 02 81 a6]

> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Private network serving the local user (1)

>                  Ext: 1  Cause: Network out of order (38), class =
Network Congestion (resource unavailable) (2) ]

    -- Hungup 'Zap/1-1'

  == Everyone is busy/congested at this time (1:0/1/0)

  == Auto fallthrough, channel 'Local/666 at test-01c4,2' status is
'CONGESTION'

  == Spawn extension (test, 665, 1) exited non-zero on
'SIP/sippuas-0823c430'

< Protocol Discriminator: Q.931 (8)  len=5

< Call Ref: len= 2 (reference 28/0x1C) (Terminator)

< Message type: RELEASE COMPLETE (90)

q931.c:3501 q931_receive: call 32796 on channel 1 enters state 0 (Null)

NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null

NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null

Von: asterisk-video-bounces at lists.digium.com
[mailto:asterisk-video-bounces at lists.digium.com] Im Auftrag von Cihan
Aksakal
Gesendet: Freitag, 2. November 2007 11:12
An: Development discussion of video media support in Asterisk
Betreff: Re: [Asterisk-video] [spam] Outbound Call / Callout Support

Hi Sergio,

Nice tool, thank you for the tip. 

But we want to make outbound video calls to 3G handsets as well. 

And I would prefer to use all components of our system during load
tests. Using only Sip would mean, that we won't use our E1 cards.

Best Regards,

Cihan

Von: asterisk-video-bounces at lists.digium.com
[mailto:asterisk-video-bounces at lists.digium.com] Im Auftrag von Sergio
Garcia Murillo
Gesendet: Donnerstag, 1. November 2007 23:13
An: Development discussion of video media support in Asterisk
Betreff: Re: [Asterisk-video] [spam] Outbound Call / Callout Support

Hi Cihan,

I used sipp (http://sipp.sourceforge.net/) to make sip load balancing
test. 

Just set up the first asterisk with a h324m_call to the second asterisk
box, it allows even to send rtp streams.

BR

Sergio

	----- Original Message ----- 

	From: Cihan Aksakal <mailto:cihan.aksakal at viif.de>  

	To: Development discussion of video media support in Asterisk
<mailto:asterisk-video at lists.digium.com>  

	Sent: Wednesday, October 31, 2007 6:28 PM

	Subject: [spam][Asterisk-video] Outbound Call / Callout Support

	Hi all,

	I am currently evaluating how to make h324m outbound calls and
need some help.

	We have two main objectives: 

	*         We want to make load tests using two asterisk machines
with up to 4 E1 ports per machine. 

	*         and outbound calls to 3G handsets.

	The outbound call should be connected to an asterisk application
which is providing a 3G-dtmf-video-portal using h324m_gw(). Inbound
calls to that application using h324m_gw() is working.

	Any ideas how to setup such an environment?

	Regards,

	Cihan Aksakal

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