[Asterisk-video] Lifesize VC and Asterisk
Olle E. Johansson
oej at edvina.net
Fri Feb 25 16:00:33 CST 2011
25 feb 2011 kl. 22.02 skrev CM Rahman:
> Yes. it is. If I dial direct, it is fine. but if I use asterisk as a sip proxy, the size is smaller. Any idea why?
1. Asterisk is not a SIP proxy, it's a b2bua
2. Asterisk strips all the properties of the video codecs making the video media useless
/O
>
> From: Jamie A. Stapleton <jstapleton at computer-business.com>
> To: Development discussion of video media support in Asterisk <asterisk-video at lists.digium.com>
> Sent: Fri, February 25, 2011 10:10:36 AM
> Subject: Re: [Asterisk-video] Lifesize VC and Asterisk
>
> Is lifesize being used on both ends of the call?
>
> On Feb 24, 2011, at 10:57 AM, CM Rahman wrote:
>
> Anybody here using asterisk and lifesize express? I am trying to use it. It dials fine but the video size is smaller. Is there any where I can twick to get the right video size?
>
> Thanks
> CM
>
> ________________________________
> From: pankaj pandey <pankaj.niet at yahoo.com<mailto:pankaj.niet at yahoo.com>>
> To: asterisk-video at lists.digium.com<mailto:asterisk-video at lists.digium.com>
> Sent: Thu, February 24, 2011 5:19:29 AM
> Subject: Re: [Asterisk-video] asterisk-video Digest, Vol 58, Issue 12
>
>
> thanks for reply Sergio...
>
> please find the attached log
>
>
>
>
>
> -- Executing [90xxxxxxxx at 3G:1] h324m_call("SIP/100-b7421e80", "90xxxxxxxx at 3Gout") in new stack
>
> -- Executing [90xxxxxxxx at 3Gout:1] Set("Local/90xxxxxxxx at 3Gout-f57d,2", "CHANNEL(transfercapability)=VIDEO") in new stack
>
> -- Executing [90xxxxxxxx at 3Gout:2] NoOp("Local/90xxxxxxxx at 3Gout-f57d,2", "transfer=VIDEO") in new stack
>
> -- Executing [90xxxxxxxx at 3Gout:3] Set("Local/90xxxxxxxx at 3Gout-f57d,2", "CHANNEL(userinformationlayer1)=38") in new stack
>
> -- Executing [90xxxxxxxx at 3Gout:4] NoOp("Local/90xxxxxxxx at 3Gout-f57d,2", "ul1=38") in new stack
>
> -- Executing [90xxxxxxxx at 3Gout:5] Dial("Local/90xxxxxxxx at 3Gout-f57d,2", "ZAP/g1/90xxxxxxxx") in new stack
>
> -- Making new call for cr 32780
>
> -- digital call, setting user information layer 1 to 38 (0x26)
>
> -- Requested transfer capability: 0x18 - VIDEO
>
> > Protocol Discriminator: Q.931 (8) len=36
>
> > Call Ref: len= 2 (reference 12/0xC) (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 (8)
>
> > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
>
> > User information layer 1: H.223 and H.245 (38)
>
> > [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 05 21 80 31 30 30]
>
> > Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>
> > Presentation: Presentation permitted, user number not screened (0) '100' ]
>
> > [70 0b 80 39 30 31 33 36 38 34 32 39 33]
>
> > Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '90xxxxxxxx' ]
>
> > [a1]ost*CLI>
>
> > Sending Complete (len= 1)
>
> q931.c:3245 q931_setup: call 32780 on channel 1 enters state 1 (Call Initiated)
>
> -- Called g1/90xxxxxxxx
>
> < Protocol Discriminator: Q.931 (8) len=10
>
> < Call Ref: len= 2 (reference 12/0xC) (Terminator)
>
> < Message type: CALL PROCEEDING (2)
>
> < [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:3800 q931_receive: call 32780 on channel 1 enters state 3 (Outgoing call Proceeding)
>
> -- Zap/1-1 is proceeding passing it to Local/90xxxxxxxx at 3Gout-f57d,2
>
> < Protocol Discriminator: Q.931 (8) len=9
>
> < Call Ref: len= 2 (reference 12/0xC) (Terminator)
>
> < Message type: PROGRESS (3)
>
> < [1e 02 8a 84]
>
> < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Network beyond the interworking point (10)
>
> < Ext: 1 Progress Description: Unknown (4) ]
>
> -- Processing IE 30 (cs0, Progress Indicator)
>
> -- Zap/1-1 is making progress passing it to Local/90xxxxxxxx at 3Gout-f57d,2
>
> < Protocol Discriminator: Q.931 (8) len=5
>
> < Call Ref: len= 2 (reference 12/0xC) (Terminator)
>
> < Message type: ALERTING (1)
>
> q931.c:3715 q931_receive: call 32780 on channel 1 enters state 4 (Call Delivered)
>
> -- Zap/1-1 is ringing
>
> < Protocol Discriminator: Q.931 (8) len=12
>
> < Call Ref: len= 2 (reference 12/0xC) (Terminator)
>
> < Message type: CONNECT (7)
>
> < [29 05 0b 02 18 0f 25]
>
> < Time Date (len= 7) [ 11-02-24 15:37 ]
>
> -- Processing IE 41 (cs0, Date/Time)
>
> q931.c:3745 q931_receive: call 32780 on channel 1 enters state 10 (Active)
>
> > Protocol Discriminator: Q.931 (8) len=5
>
> > Call Ref: len= 2 (reference 12/0xC) (Originator)
>
> > Message type: CONNECT ACKNOWLEDGE (15)
>
> -- Zap/1-1 answered Local/90xxxxxxxx at 3Gout-f57d,2
>
> == Spawn extension (3Gout, 90xxxxxxxx, 5) exited non-zero on 'Local/90xxxxxxxx at 3Gout-f57d,2'
>
> < Protocol Discriminator: Q.931 (8) len=9
>
> < Call Ref: len= 2 (reference 12/0xC) (Terminator)
>
> < Message type: DISCONNECT (69)
>
> < [08 02 80 90]
>
> < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0)
>
> < Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]
>
> -- Processing IE 8 (cs0, Cause)
>
> q931.c:3935 q931_receive: call 32780 on channel 1 enters state 12 (Disconnect Indication)
>
> -- Channel 0/1, span 1 got hangup request, cause 16
>
> NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request
>
> q931.c:3068 q931_release: call 32780 on channel 1 enters state 19 (Release Request)
>
> > Protocol Discriminator: Q.931 (8) len=9
>
> > Call Ref: len= 2 (reference 12/0xC) (Originator)
>
> > Message type: RELEASE (77)
>
> > [08 02 81 90]
>
> > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1)
>
> > Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ]
>
> -- Hungup 'Zap/1-1'
>
> == Auto fallthrough, channel 'SIP/100-b7421e80' status is 'UNKNOWN'
>
> < Protocol Discriminator: Q.931 (8) len=5
>
> < Call Ref: len= 2 (reference 12/0xC) (Terminator)
>
> < Message type: RELEASE COMPLETE (90)
>
> q931.c:3875 q931_receive: call 32780 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
>
>
> --- On Thu, 24/2/11, asterisk-video-request at lists.digium.com<mailto:asterisk-video-request at lists.digium.com> <asterisk-video-request at lists.digium.com<mailto:asterisk-video-request at lists.digium.com>> wrote:
>
> From: asterisk-video-request at lists.digium.com<mailto:asterisk-video-request at lists.digium.com> <asterisk-video-request at lists.digium.com<mailto:asterisk-video-request at lists.digium.com>>
> Subject: asterisk-video Digest, Vol 58, Issue 12
> To: asterisk-video at lists.digium.com<mailto:asterisk-video at lists.digium.com>
> Date: Thursday, 24 February, 2011, 3:48 AM
>
> Send asterisk-video mailing list submissions to
> asterisk-video at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-video
> or, via email, send a message with subject or body 'help' to
> asterisk-video-request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-video-owner at lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-video digest..."
>
>
> Today's Topics:
>
> 1. Re: video obd call |h324m gw (sudhir mor)
> 2. Re: video obd call |h324m gw (Sergio Garcia Murillo)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 24 Feb 2011 13:57:57 +0530 (IST)
> From: sudhir mor <sudhir_mor2000 at yahoo.com>
> Subject: Re: [Asterisk-video] video obd call |h324m gw
> To: Development discussion of video media support in Asterisk
> <asterisk-video at lists.digium.com>
> Message-ID: <711770.75556.qm at web94816.mail.in2.yahoo.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Pankaj,
>
> Please follow help from this link
> https://issues.asterisk.org/view.php?id=10189
> ?
> Sudhir Mor
> Senior Developer
> Voicetap Technologies
> Mobile : +91-9891318796
> ________________________________
>
>
>
>
>
> ________________________________
> From: pankaj pandey <pankaj.niet at yahoo.com>
> To: asterisk-video at lists.digium.com
> Sent: Thu, 24 February, 2011 1:35:02 PM
> Subject: [Asterisk-video] video obd call |h324m gw
>
>
> Hi everyone,
> ?
> My first scenario
> 3G phone -> asterisk(h324m gw)->sip
> Is working fine.
> ?
> when I try a video OBD from sip
> i.e.
> SIP -> asterisk(h324m gw)-> 3G phone
> ?
> Video OBD call is originated at 3G phone end and it is shows as video call, but
> when I picking the call it shows an ?Unknown Error? and call cut with ?hangup
> request, cause 16..
> ?
> below is the dial-plan and cli log.
> ?
> ?
> please suggest the way forward...
> ?
> ?
> ?
> [3G]
> exten =>? _X.,1,h324m_call(${EXTEN}@3Gout)
> ?
> [3Gout]
> exten =>? _X.,1,Set(CHANNEL(transfercapability)=VIDEO)
> exten =>? _X.,2,NoOp(transfer=${CHANNEL(transfercapability)})
> exten =>? _X.,3,Set(CHANNEL(userinformationlayer1)=38)
> exten =>? _X.,4,NoOp(ul1=${CHANNEL(userinformationlayer1)})
> exten =>? _X.,n,Dial(ZAP/g1/${EXTEN})
> ?
> ?
> - Executing [93xxxxxxxx at 3G:1] h324m_call("SIP/100-096dc4a0", "93xxxxxxxx at 3Gout")
> in new stack
> ??? -- Executing [93xxxxxxxx at 3Gout:1] Set("Local/93xxxxxxxx at 3Gout-ad7c,2",
> "CHANNEL(transfercapability)=VIDEO") in new stack
> ??? -- Executing [93xxxxxxxx at 3Gout:2] NoOp("Local/93xxxxxxxx at 3Gout-ad7c,2",
> "transfer=VIDEO") in new stack
> ??? -- Executing [93xxxxxxxx at 3Gout:3] Set("Local/93xxxxxxxx at 3Gout-ad7c,2",
> "CHANNEL(userinformationlayer1)=38") in new stack
> ??? -- Executing [93xxxxxxxx at 3Gout:4] NoOp("Local/93xxxxxxxx at 3Gout-ad7c,2",
> "ul1=38") in new stack
> ??? -- Executing [93xxxxxxxx at 3Gout:5] Dial("Local/93xxxxxxxx at 3Gout-ad7c,2",
> "ZAP/g1/93xxxxxxxx") in new stack
> ??? -- digital call, setting user information layer 1 to 38 (0x26)
> ??? -- Requested transfer capability: 0x18 - VIDEO
> ??? -- Called g1/93xxxxxxxx
> ??? -- Zap/1-1 is proceeding passing it to Local/93xxxxxxxx at 3Gout-ad7c,2
> ??? -- Zap/1-1 is making progress passing it to Local/93xxxxxxxx at 3Gout-ad7c,2
> ??? -- Zap/1-1 is ringing
> ??? -- Zap/1-1 answered Local/93xxxxxxxx at 3Gout-ad7c,2
> ? == Spawn extension (3Gout, 93xxxxxxxx, 5) exited non-zero on
> 'Local/93xxxxxxxx at 3Gout-ad7c,2'
> ??? -- Channel 0/1, span 1 got hangup request, cause 16
> ??? -- Hungup 'Zap/1-1'
>
> Thanks,
> Pankaj
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://lists.digium.com/pipermail/asterisk-video/attachments/20110224/b6308506/attachment-0001.htm>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 24 Feb 2011 09:47:00 +0100
> From: Sergio Garcia Murillo <sergio.garcia at fontventa.com>
> Subject: Re: [Asterisk-video] video obd call |h324m gw
> To: Development discussion of video media support in Asterisk
> <asterisk-video at lists.digium.com>
> Message-ID: <4D661B04.3080300 at fontventa.com>
> Content-Type: text/plain; charset="utf-8"; Format="flowed"
>
>
> Enable debug on asterisk and attach log again
>
> Best regards
> Sergio
>
> El 24/02/2011 9:05, pankaj pandey escribi?:
> >
> > Hi everyone,
> >
> > My first scenario
> >
> > 3G phone -> asterisk(h324m gw)->sip
> >
> > Is working fine.
> >
> > when I try a video OBD from sip
> >
> > i.e.
> >
> > SIP -> asterisk(h324m gw)-> 3G phone
> >
> > Video OBD call is originated at 3G phone end and it is shows as video
> > call, but when I picking the call it shows an ?Unknown Error? and call
> > cut with hangup request, cause 16..
> >
> > below is the dial-plan and cli log.
> >
> > please suggest the way forward...
> >
> > [3G]
> >
> > exten =>_X.,1,h324m_call(${EXTEN}@3Gout)
> >
> > [3Gout]
> >
> > exten =>_X.,1,Set(CHANNEL(transfercapability)=VIDEO)
> >
> > exten =>_X.,2,NoOp(transfer=${CHANNEL(transfercapability)})
> >
> > exten =>_X.,3,Set(CHANNEL(userinformationlayer1)=38)
> >
> > exten =>_X.,4,NoOp(ul1=${CHANNEL(userinformationlayer1)})
> >
> > exten =>_X.,n,Dial(ZAP/g1/${EXTEN})
> >
> > - Executing [93xxxxxxxx at 3G:1] h324m_call("SIP/100-096dc4a0",
> > "93xxxxxxxx at 3Gout") in new stack
> >
> > -- Executing [93xxxxxxxx at 3Gout:1] Set("Local/93xxxxxxxx at 3Gout-ad7c,2",
> > "CHANNEL(transfercapability)=VIDEO") in new stack
> >
> > -- Executing [93xxxxxxxx at 3Gout:2]
> > NoOp("Local/93xxxxxxxx at 3Gout-ad7c,2", "transfer=VIDEO") in new stack
> >
> > -- Executing [93xxxxxxxx at 3Gout:3] Set("Local/93xxxxxxxx at 3Gout-ad7c,2",
> > "CHANNEL(userinformationlayer1)=38") in new stack
> >
> > -- Executing [93xxxxxxxx at 3Gout:4]
> > NoOp("Local/93xxxxxxxx at 3Gout-ad7c,2", "ul1=38") in new stack
> >
> > -- Executing [93xxxxxxxx at 3Gout:5]
> > Dial("Local/93xxxxxxxx at 3Gout-ad7c,2", "ZAP/g1/93xxxxxxxx") in new stack
> >
> > -- digital call, setting user information layer 1 to 38 (0x26)
> >
> > -- Requested transfer capability: 0x18 - VIDEO
> >
> > -- Called g1/93xxxxxxxx
> >
> > -- Zap/1-1 is proceeding passing it to Local/93xxxxxxxx at 3Gout-ad7c,2
> >
> > -- Zap/1-1 is making progress passing it to Local/93xxxxxxxx at 3Gout-ad7c,2
> >
> > -- Zap/1-1 is ringing
> >
> > -- Zap/1-1 answered Local/93xxxxxxxx at 3Gout-ad7c,2
> >
> > == Spawn extension (3Gout, 93xxxxxxxx, 5) exited non-zero on
> > 'Local/93xxxxxxxx at 3Gout-ad7c,2'
> >
> > -- Channel 0/1, span 1 got hangup request, cause 16
> >
> > -- Hungup 'Zap/1-1'
> >
> >
> >
> > Thanks,
> > Pankaj
> >
> >
> >
> > --
> > _____________________________________________________________________
> > -- 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
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://lists.digium.com/pipermail/asterisk-video/attachments/20110224/bb14a1fb/attachment.htm>
>
> ------------------------------
>
> _______________________________________________
> --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
>
> End of asterisk-video Digest, Vol 58, Issue 12
> **********************************************
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
> --
> _____________________________________________________________________
> -- 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
---
* Olle E Johansson - oej at edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
More information about the asterisk-video
mailing list