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[asterisk-users] Fw: Outbound video Calls


 
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PostPosted: Thu Jun 26, 2008 6:55 am    Post subject: [asterisk-users] Fw: Outbound video Calls Reply with quote

Quote:
Hi,

Quote:
You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from
http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too.
Maybe the switch wants to have it in Bearer Capability and LCC (I once
had such a switch).


Just applied the patch, failed again. can you tell me if theres anything
more i need to add to the conf file to signal in LLC as well ?


Quote:
Another reason could be that the telco blocks video calls.


They keep telling me that there shouldnt be a problem, however they are
not the brightest bunch Smile


Quote:
regards
klaus

PS: use the asterisk-video mailing lists

Just have Smile



Quote:

Asterisk Users schrieb:
Quote:
Hi all,

I am trying to make an outbound video call to a mobile from asterisk.
however it keeps failing.

I can make inbound calls from a mobile and view video.
I am using x-lite to initiate the outbound call, however I have tried
using
the management interface as well (action: etc...) and result is the
same.

normal voice outbound calls work fine.

Circuit is a q931 30 channel from telewest (virgin media).

Any pointers would be appreciated.

below is pri debug output and relevant conf entries.

// BEGIN //

-- Executing [666 at sip_in:1] Goto("SIP/paul-081ff260",
"video_test_out|666|1") in new stack

-- Goto (video_test_out,666,1)

-- Executing [666 at video_test_out:1] Set("SIP/paul-081ff260",
"CHANNEL(transfercapability)=VIDEO") in new stack

-- Executing [666 at video_test_out:2] Set("SIP/paul-081ff260",
"CHANNEL(userinformationlayer1)=38") in new stack

-- Executing [666 at video_test_out:3] h324m_gw("SIP/paul-081ff260",
"s at video_test_out_context") in new stack

[Jun 26 09:21:46] WARNING[7881]: channel.c:700 ast_best_codec: Don't
know
any of 0x2000 formats

-- Executing [s at video_test_out_context:1]
h324m_call("Local/s at video_test_out_context-f51e,2",
"dialcell at video_test_out_context") in new stack

-- Executing [dialcell at video_test_out_context:1]
Set("Local/dialcell at video_test_out_context-de13,2",
"CHANNEL(transfercapability)=VIDEO") in new stack

-- Executing [dialcell at video_test_out_context:2]
NoOp("Local/dialcell at video_test_out_context-de13,2", "transfer=VIDEO")
in
new stack

-- Executing [dialcell at video_test_out_context:3]
Set("Local/dialcell at video_test_out_context-de13,2",
"CHANNEL(userinformationlayer1)=38") in new stack

-- Executing [dialcell at video_test_out_context:4]
NoOp("Local/dialcell at video_test_out_context-de13,2", "ul1=38") in new
stack

-- Executing [dialcell at video_test_out_context:5]
Dial("Local/dialcell at video_test_out_context-de13,2",
"Zap/g0/07525029025|40|tTkK") in new stack

-- Making new call for cr 32771

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

-- Requested transfer capability: 0x18 - VIDEO

Quote:
Protocol Discriminator: Q.931 (Cool len=38

Quote:
Call Ref: len= 2 (reference 3/0x3) (Originator)

Quote:
Message type: SETUP (5)

Quote:
[04 03 88 90 a6]

Quote:
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Unrestricted digital information (Cool

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

Quote:
Ext: 1 User information layer 1: H.223
and
H.245 (3Cool

Quote:
[18 03 a9 83 81]

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

Quote:
ChanSel: Reserved

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

Quote:
Ext: 1 Channel: 1 ]

Quote:
[6c 06 41 80 70 61 75 6c]

Quote:
Calling Number (len= Cool [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)

Quote:
Presentation: Presentation permitted, user
number not screened (0) 'paul' ]

Quote:
[70 0c c1 30 37 35 32 35 30 32 39 30 32 35]

Quote:
Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '07525029025' ]

Quote:
[a1]CLI>

Quote:
Sending Complete (len= 1)

q931.c:2881 q931_setup: call 32771 on channel 1 enters state 1 (Call
Initiated)

-- Called g0/07525029025

< Protocol Discriminator: Q.931 (Cool len=10

< Call Ref: len= 2 (reference 3/0x3) (Terminator)

< Message type: RELEASE COMPLETE (90)

< [08 03 80 e4 04]

< Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
Location: User (0)

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

< Cause data 1: 04 (4)

-- Processing IE 8 (cs0, Cause)

q931.c:3503 q931_receive: call 32771 on channel 1 enters state 0 (Null)

-- Channel 0/1, span 1 got hangup, cause 100

NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null

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 [dialcell at video_test_out_context:6]
Hangup("Local/dialcell at video_test_out_context-de13,2", "") in new stack

== Spawn extension (video_test_out_context, dialcell, 6) exited
non-zero
on 'Local/dialcell at video_test_out_context-de13,2'

== Auto fallthrough, channel 'Local/s at video_test_out_context-f51e,2'
status is 'UNKNOWN'

== Spawn extension (video_test_out, 666, 3) exited non-zero on
'SIP/paul-081ff260'



// END //





extensions.conf:



[video_test_out]

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

exten => 666,n,Set(CHANNEL(userinformationlayer1)=3Cool

exten => 666,n,h324m_gw(s at video_test_out_context)

exten => 666,n,Hangup



[video_test_out_context]

exten => s,1,h324m_call(dialcell at video_test_out_context)

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

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

exten => dialcell,n,Set(CHANNEL(userinformationlayer1)=3Cool

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

exten => dialcell,n,Dial(Zap/g0/07xxxxxxxxx,40,tTkK)

exten => dialcell,n,Hangup()

exten => t,1,Goto(s,2)



sip.conf:



[general]

context=sip_in

allowoverlap=no

bindport=5060

bindaddr=0.0.0.0

videosupport=yes

disable=all

allow=ulaw

allow=alaw

allow=h263+

;allow=h263

;allow=h263p

allow=speex

allow=gsm

#include "/etc/pbx-tandil/sip.conf"

#include "/etc/asterisk/sip_dps.conf"



[paul]

type=friend

username=paul

secret=georgina

nat=never

host=dynamic

canreinvite=no

allow=h263p

--

Paul Verity







_______________________________________________
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klaus.mailinglists at ...
Guest





PostPosted: Thu Jun 26, 2008 9:36 am    Post subject: [asterisk-users] Fw: Outbound video Calls Reply with quote

you also need (as stated in the bug report) the patch
10217-asterisk-unrestricted-digital-llc-11595-1.4.17.patch from
http://bugs.digium.com/view.php?id=10217

This enables LCC in chan_zap. Is this was done some time ago I do not
remember anymore who it is activated, I think you have to add the
h324m=lcc
option to zapata.conf

I remember one scenario where H324M signaling was required to be in
Bearer Capabilite AND Low Layer Compatibility. I think you can easily
extend the patches to signal both versions at the same time.

Always take a look at the outgoing SETUP message to see if it contains LCC.

PS: Please dump an incoming SETUP message for a video call - does it
contain LCC too?

regards
klaus

Asterisk Users schrieb:
Quote:
Quote:
Hi,

Quote:
You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from
http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too.
Maybe the switch wants to have it in Bearer Capability and LCC (I once
had such a switch).

Just applied the patch, failed again. can you tell me if theres anything
more i need to add to the conf file to signal in LLC as well ?


Quote:
Another reason could be that the telco blocks video calls.

They keep telling me that there shouldnt be a problem, however they are
not the brightest bunch Smile


Quote:
regards
klaus

PS: use the asterisk-video mailing lists
Just have Smile



Quote:
Asterisk Users schrieb:
Quote:
Hi all,

I am trying to make an outbound video call to a mobile from asterisk.
however it keeps failing.

I can make inbound calls from a mobile and view video.
I am using x-lite to initiate the outbound call, however I have tried
using
the management interface as well (action: etc...) and result is the
same.

normal voice outbound calls work fine.

Circuit is a q931 30 channel from telewest (virgin media).

Any pointers would be appreciated.

below is pri debug output and relevant conf entries.

// BEGIN //

-- Executing [666 at sip_in:1] Goto("SIP/paul-081ff260",
"video_test_out|666|1") in new stack

-- Goto (video_test_out,666,1)

-- Executing [666 at video_test_out:1] Set("SIP/paul-081ff260",
"CHANNEL(transfercapability)=VIDEO") in new stack

-- Executing [666 at video_test_out:2] Set("SIP/paul-081ff260",
"CHANNEL(userinformationlayer1)=38") in new stack

-- Executing [666 at video_test_out:3] h324m_gw("SIP/paul-081ff260",
"s at video_test_out_context") in new stack

[Jun 26 09:21:46] WARNING[7881]: channel.c:700 ast_best_codec: Don't
know
any of 0x2000 formats

-- Executing [s at video_test_out_context:1]
h324m_call("Local/s at video_test_out_context-f51e,2",
"dialcell at video_test_out_context") in new stack

-- Executing [dialcell at video_test_out_context:1]
Set("Local/dialcell at video_test_out_context-de13,2",
"CHANNEL(transfercapability)=VIDEO") in new stack

-- Executing [dialcell at video_test_out_context:2]
NoOp("Local/dialcell at video_test_out_context-de13,2", "transfer=VIDEO")
in
new stack

-- Executing [dialcell at video_test_out_context:3]
Set("Local/dialcell at video_test_out_context-de13,2",
"CHANNEL(userinformationlayer1)=38") in new stack

-- Executing [dialcell at video_test_out_context:4]
NoOp("Local/dialcell at video_test_out_context-de13,2", "ul1=38") in new
stack

-- Executing [dialcell at video_test_out_context:5]
Dial("Local/dialcell at video_test_out_context-de13,2",
"Zap/g0/07525029025|40|tTkK") in new stack

-- Making new call for cr 32771

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

-- Requested transfer capability: 0x18 - VIDEO

Quote:
Protocol Discriminator: Q.931 (Cool len=38
Call Ref: len= 2 (reference 3/0x3) (Originator)
Message type: SETUP (5)
[04 03 88 90 a6]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Unrestricted digital information (Cool
Ext: 1 Trans mode/rate: 64kbps,
circuit-mode
(16)
Ext: 1 User information layer 1: H.223
and
H.245 (3Cool
[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 41 80 70 61 75 6c]
Calling Number (len= Cool [ Ext: 0 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
number not screened (0) 'paul' ]
[70 0c c1 30 37 35 32 35 30 32 39 30 32 35]
Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '07525029025' ]
[a1]CLI>
Sending Complete (len= 1)
q931.c:2881 q931_setup: call 32771 on channel 1 enters state 1 (Call
Initiated)

-- Called g0/07525029025

< Protocol Discriminator: Q.931 (Cool len=10

< Call Ref: len= 2 (reference 3/0x3) (Terminator)

< Message type: RELEASE COMPLETE (90)

< [08 03 80 e4 04]

< Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0
Location: User (0)

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

< Cause data 1: 04 (4)

-- Processing IE 8 (cs0, Cause)

q931.c:3503 q931_receive: call 32771 on channel 1 enters state 0 (Null)

-- Channel 0/1, span 1 got hangup, cause 100

NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null

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 [dialcell at video_test_out_context:6]
Hangup("Local/dialcell at video_test_out_context-de13,2", "") in new stack

== Spawn extension (video_test_out_context, dialcell, 6) exited
non-zero
on 'Local/dialcell at video_test_out_context-de13,2'

== Auto fallthrough, channel 'Local/s at video_test_out_context-f51e,2'
status is 'UNKNOWN'

== Spawn extension (video_test_out, 666, 3) exited non-zero on
'SIP/paul-081ff260'



// END //





extensions.conf:



[video_test_out]

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

exten => 666,n,Set(CHANNEL(userinformationlayer1)=3Cool

exten => 666,n,h324m_gw(s at video_test_out_context)

exten => 666,n,Hangup



[video_test_out_context]

exten => s,1,h324m_call(dialcell at video_test_out_context)

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

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

exten => dialcell,n,Set(CHANNEL(userinformationlayer1)=3Cool

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

exten => dialcell,n,Dial(Zap/g0/07xxxxxxxxx,40,tTkK)

exten => dialcell,n,Hangup()

exten => t,1,Goto(s,2)



sip.conf:



[general]

context=sip_in

allowoverlap=no

bindport=5060

bindaddr=0.0.0.0

videosupport=yes

disable=all

allow=ulaw

allow=alaw

allow=h263+

;allow=h263

;allow=h263p

allow=speex

allow=gsm

#include "/etc/pbx-tandil/sip.conf"

#include "/etc/asterisk/sip_dps.conf"



[paul]

type=friend

username=paul

secret=georgina

nat=never

host=dynamic

canreinvite=no

allow=h263p

--

Paul Verity







_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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