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 <[email protected]> To: [email protected] 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@3G:1] h324m_call("SIP/100-b7421e80", "90xxxxxxxx@3Gout") in new stack -- Executing [90xxxxxxxx@3Gout:1] Set("Local/90xxxxxxxx@3Gout-f57d,2", "CHANNEL(transfercapability)=VIDEO") in new stack -- Executing [90xxxxxxxx@3Gout:2] NoOp("Local/90xxxxxxxx@3Gout-f57d,2", "transfer=VIDEO") in new stack -- Executing [90xxxxxxxx@3Gout:3] Set("Local/90xxxxxxxx@3Gout-f57d,2", "CHANNEL(userinformationlayer1)=38") in new stack -- Executing [90xxxxxxxx@3Gout:4] NoOp("Local/90xxxxxxxx@3Gout-f57d,2", "ul1=38") in new stack -- Executing [90xxxxxxxx@3Gout:5] Dial("Local/90xxxxxxxx@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@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@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@3Gout-f57d,2 == Spawn extension (3Gout, 90xxxxxxxx, 5) exited non-zero on 'Local/90xxxxxxxx@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, [email protected] <[email protected]> wrote: >From: [email protected] ><[email protected]> >Subject: asterisk-video Digest, Vol 58, Issue 12 >To: [email protected] >Date: Thursday, 24 February, 2011, 3:48 AM > > >Send asterisk-video mailing list submissions to > [email protected] > >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 > [email protected] > >You can reach the person managing the list at > [email protected] > >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 <[email protected]> >Subject: Re: [Asterisk-video] video obd call |h324m gw >To: Development discussion of video media support in Asterisk > <[email protected]> >Message-ID: <[email protected]> >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 <[email protected]> >To: [email protected] >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@3G:1] h324m_call("SIP/100-096dc4a0", >"93xxxxxxxx@3Gout") > >in new stack >??? -- Executing [93xxxxxxxx@3Gout:1] Set("Local/93xxxxxxxx@3Gout-ad7c,2", >"CHANNEL(transfercapability)=VIDEO") in new stack >??? -- Executing [93xxxxxxxx@3Gout:2] NoOp("Local/93xxxxxxxx@3Gout-ad7c,2", >"transfer=VIDEO") in new stack >??? -- Executing [93xxxxxxxx@3Gout:3] Set("Local/93xxxxxxxx@3Gout-ad7c,2", >"CHANNEL(userinformationlayer1)=38") in new stack >??? -- Executing [93xxxxxxxx@3Gout:4] NoOp("Local/93xxxxxxxx@3Gout-ad7c,2", >"ul1=38") in new stack >??? -- Executing [93xxxxxxxx@3Gout:5] Dial("Local/93xxxxxxxx@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@3Gout-ad7c,2 >??? -- Zap/1-1 is making progress passing it to Local/93xxxxxxxx@3Gout-ad7c,2 >??? -- Zap/1-1 is ringing >??? -- Zap/1-1 answered Local/93xxxxxxxx@3Gout-ad7c,2 >? == Spawn extension (3Gout, 93xxxxxxxx, 5) exited non-zero on >'Local/93xxxxxxxx@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 <[email protected]> >Subject: Re: [Asterisk-video] video obd call |h324m gw >To: Development discussion of video media support in Asterisk > <[email protected]> >Message-ID: <[email protected]> >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@3G:1] h324m_call("SIP/100-096dc4a0", >> "93xxxxxxxx@3Gout") in new stack >> >> -- Executing [93xxxxxxxx@3Gout:1] Set("Local/93xxxxxxxx@3Gout-ad7c,2", >> "CHANNEL(transfercapability)=VIDEO") in new stack >> >> -- Executing [93xxxxxxxx@3Gout:2] >> NoOp("Local/93xxxxxxxx@3Gout-ad7c,2", "transfer=VIDEO") in new stack >> >> -- Executing [93xxxxxxxx@3Gout:3] Set("Local/93xxxxxxxx@3Gout-ad7c,2", >> "CHANNEL(userinformationlayer1)=38") in new stack >> >> -- Executing [93xxxxxxxx@3Gout:4] >> NoOp("Local/93xxxxxxxx@3Gout-ad7c,2", "ul1=38") in new stack >> >> -- Executing [93xxxxxxxx@3Gout:5] >> Dial("Local/93xxxxxxxx@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@3Gout-ad7c,2 >> >> -- Zap/1-1 is making progress passing it to Local/93xxxxxxxx@3Gout-ad7c,2 >> >> -- Zap/1-1 is ringing >> >> -- Zap/1-1 answered Local/93xxxxxxxx@3Gout-ad7c,2 >> >> == Spawn extension (3Gout, 93xxxxxxxx, 5) exited non-zero on >> 'Local/93xxxxxxxx@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 >********************************************** >
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