Re: EX90 and CUCM

anthonymd · ‎03-26-2012

Hello,

I have a couple of EX90s here that I want to get dialed in. I can connect and do video conferencing with all of our vid conf devices. The thing I'm trying to do is get incoming calls from the PSTN to ring this thing. I can get it to ring by setting it up as a remote destination device as part of SNR but when I pick up it immediately hangs up the call and my other devices continue to ring. I have a feeling it's a transcoding issue but can't find where. I am able to dial from the EX90 to other IP phones on the network and vice versa. Any ideas would be gratefully accepted.

mubakhta · ‎03-26-2012

Hi Anthony,

My name is Mubashshir and I am from the Telepresence Support team.

Is it possible for you to enable the debugging on the Ex90 so that we can monitor the exchange of messages which are received? Also, it would be great if you can describe the network you have setup for PSTN calls.

Then there are some basic questions which you can answer:

-Is your Ex90 registered directly to the CUCM or it is registered to another VCS?

-If your endpoint is registered to the VCS, do you have a SIP trunk in the CUCM for the VCS?

Our main concern is to find out the reason of disconnection. Enabling the debugging, we should be able to figure that out.

Regards,

Mubashshir Akhtar

Thanks Mubashshir Akhtar

anthonymd · ‎03-26-2012

Regards,

Tony D'Astolfo

Network Engineer

Sierra Nevada Corporation

www.sncorp.com

Main: 775-331-0222 x216886

Direct: 775-849-6886

From: mubakhta

To: anthonymd

Date: 03/26/2012 03:06 PM

Subject: - Re: EX90 and CUCM

Home

Re: EX90 and CUCM

created by Mubashshir Akhtar in TelePresence - View the full discussion

Hi Anthony,

My name is Mubashshir and I am from the Telepresence Support team.

Is it possible for you to enable the debugging on the Ex90 so that we can

monitor the exchange of messages which are received? Also, it would be

great if you can describe the network you have setup for PSTN calls.

PSTN to Cisco router via PRIs, to CM, to VCS via SIP trunk. Pretty basic

setup really.

Then there are some basic questions which you can answer:

-Is your Ex90 registered directly to the CUCM or it is registered to

another VCS?

Registered to VCS

-If your endpoint is registered to the VCS, do you have a SIP trunk in the

CUCM for the VCS?

Yes

Our main concern is to find out the reason of disconnection. Enabling the

debugging, we should be able to figure that out.

Regards,

Mubashshir Akhtar

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anthonymd · ‎03-26-2012

Also, how do you enable debugging on the EX90?

Regards,

Tony D'Astolfo

Network Engineer

Sierra Nevada Corporation

www.sncorp.com

Main: 775-331-0222 x216886

Direct: 775-849-6886

From: mubakhta

To: anthonymd

Date: 03/26/2012 03:06 PM

Subject: - Re: EX90 and CUCM

Home

Re: EX90 and CUCM

created by Mubashshir Akhtar in TelePresence - View the full discussion

Hi Anthony,

My name is Mubashshir and I am from the Telepresence Support team.

Is it possible for you to enable the debugging on the Ex90 so that we can

monitor the exchange of messages which are received? Also, it would be

great if you can describe the network you have setup for PSTN calls.

Then there are some basic questions which you can answer:

-Is your Ex90 registered directly to the CUCM or it is registered to

another VCS?

-If your endpoint is registered to the VCS, do you have a SIP trunk in the

CUCM for the VCS?

Our main concern is to find out the reason of disconnection. Enabling the

debugging, we should be able to figure that out.

Regards,

Mubashshir Akhtar

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Start a new discussion in TelePresence at Home

vivsing2 · ‎03-26-2012

Hi Anthony,

You can use following method to enable log:

Commands in bold

 Open the console/telnet/ssh session with codec

 Reboot codec by xCommand boot

 Reopen the console/telnet/ssh session with codec

 xstatus

 xconfig

 log ctx SipPacket debug 9

 Make a call and keep running until you have recreated the problem

 xstatus

 Hang up call

 log ctx SipPacket debug off

 Once recreated the problem and captured the log, reboot codec by xCommand boot

 Access to codec WebGUI for download log file, http://

 Download the latest file “log.tar.gz”

 Attach file to this discussion

Regards,

Vivek

anthonymd · ‎03-27-2012

Hello Vivek,

Thank you for your help. The log command is not working. I get the

following:

log ctx SipPacket debug 9

Command not recognized.

From: vivsing2

To: anthonymd

Date: 03/26/2012 03:31 PM

Subject: - Re: EX90 and CUCM

Home

Re: EX90 and CUCM

created by Vivek Singh in TelePresence - View the full discussion

Hi Anthony,

You can use following method to enable log:

Commands in bold

� Open the console/telnet/ssh session with codec

� Reboot codec by xCommand boot

� Reopen the console/telnet/ssh session with codec

� xstatus

� xconfig

� log ctx SipPacket debug 9

� Make a call and keep running until you have recreated the problem

� xstatus

� Hang up call

� log ctx SipPacket debug off

� Once recreated the problem and captured the log, reboot codec by

xCommand boot

� Access to codec WebGUI for download log file, http://

� Download the latest file “log.tar.gz”

� Attach file to this discussion

Regards,

Vivek

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mubakhta · ‎03-26-2012

Hi Anthony,

You can use the commands given by Vivek. Also, if you just want to paste the output, please make the following changes :

Commands in bold

Make sure logging is enabled on the Putty or the terminal software you are using
Open the console/telnet/ssh session with codec
Reboot codec by xCommand boot
Reopen the console/telnet/ssh session with codec
xstatus
xconfig
•· log ctx SipPacket debug 9
log output on
Make a call and keep running until you have recreated the problem
•· xstatus
Hang up call
•· log ctx SipPacket debug off
log output off

Copy and paste the call logs in this discussion from the log text file.

Thanks Mubashshir Akhtar

mubakhta · ‎03-27-2012

Hi Anthony,

Can you make sure that you are running the log command on the Ex90 in the API interface?

You should be logged in as admin and then try to run this command on the Ex90. Also, I had mentioned about the Media Termination Point.

Can you verify the setting of the Media Termination Point on the CUCM in the phone profiles and on the SIP Trunk?

Regards,

Mubashshir Akhtar

Thanks Mubashshir Akhtar

anthonymd · ‎03-27-2012

Please find below the output you requested.

As for the MTP setting, none are set.

2412.11 SipPacket PacketDump: Proto: SIP, Name: INVITE

sip:217886@10.2.15.177:5061;transport=tls SIP/2.0, Direction: Incoming,

RemoteAddress: 10.2.2.27:5061, GroupEntity:

1df72600-f711e965-e809e-b02020a@10.2.2.11, Time: 2412107, Content: !!!<

2412.11 SipPacket INVITE sip:217886@10.2.15.177:5061;transport=tls

SIP/2.0

2412.11 SipPacket Via: SIP/2.0/TLS

10.2.2.27:5061;egress-zone=DefaultSubZone;branch=z9hG4bK53462630b7e5546a2c7fbf752eed6e6c1467.8830a1b6a042c47e99b7857db0440f2d;proxy-call-id=1f872884-7829-11e1-b44a-0010f31f232e;rport

2412.11 SipPacket Via: SIP/2.0/TCP

10.2.2.11:5060;branch=z9hG4bK11bd4d4ccd339d;received=10.2.2.11;ingress-zone=CUCM

2412.11 SipPacket Call-ID: 1df72600-f711e965-e809e-b02020a@10.2.2.11

2412.11 SipPacket CSeq: 101 INVITE

2412.11 SipPacket Remote-Party-ID: "Tony D'Astolfo"

<216886>;party=calling;screen=yes;privacy=off

2412.11 SipPacket Contact: <216886>

2412.11 SipPacket From: "Tony D'Astolfo"

<216886>;tag=2628263~672ae68a-0b27-4f6e-b42e-35e9e0bc74c3-54922102

2412.11 SipPacket To: <217886>

2412.11 SipPacket Max-Forwards: 69

2412.11 SipPacket Record-Route:

2412.11 SipPacket Allow:

INVITE,OPTIONS,INFO,BYE,CANCEL,ACK,PRACK,REFER,SUBSCRIBE,NOTIFY

2412.11 SipPacket User-Agent: Cisco-CUCM8.5

2412.11 SipPacket Expires: 180

2412.11 SipPacket Date: Tue, 27 Mar 2012 16:23:01 GMT

2412.11 SipPacket Supported:

timer,resource-priority,replaces,X-cisco-srtp-fallback,Geolocation

2412.11 SipPacket Session-Expires: 1800

2412.11 SipPacket Min-SE: 1800

2412.12 SipPacket Allow-Events: presence,kpml

2412.12 SipPacket X-TAATag: 1f872960-7829-11e1-becd-0010f31f232e

2412.12 SipPacket Call-Info:

<10.2.2.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"

2412.12 SipPacket Cisco-Guid:

0502736384-0000065536-0000017264-0184680970

2412.12 SipPacket Content-Length: 0

2412.12 SipPacket

2412.12 SipPacket >!!!

2412.12 SipStack I: SipUa(ind=-1) SIPUA_handleRecvRequest 3

2412.13 SipCall I: sip_call_handler::handleSIPMCallInd(na/0/0): Incoming

callrate=0 to sip:217886@10.2.2.27 from sip:216886@10.2.2.11

2412.13 SipCall I: sip_call::outgoingAlertIndication(): Sending alert

indication

2412.14 CAMERA I: CamVisca::Ready_doCAMActionReq cameraId=1 actionId=19

2412.15 SipStack I: SipUa GRUU added OK

2412.15 SipPacket PacketDump: Proto: SIP, Name: SIP/2.0 180 Ringing,

Direction: Outgoing, RemoteAddress: 10.2.2.27:5061, GroupEntity:

1df72600-f711e965-e809e-b02020a@10.2.2.11, Time: 2412152, Content: !!!<

2412.15 SipPacket SIP/2.0 180 Ringing

2412.16 SipPacket Via: SIP/2.0/TLS

10.2.2.27:5061;egress-zone=DefaultSubZone;branch=z9hG4bK53462630b7e5546a2c7fbf752eed6e6c1467.8830a1b6a042c47e99b7857db0440f2d;proxy-call-id=1f872884-7829-11e1-b44a-0010f31f232e;received=10.2.2.27;rport=5061

2412.16 SipPacket Via: SIP/2.0/TCP

10.2.2.11:5060;branch=z9hG4bK11bd4d4ccd339d;received=10.2.2.11;ingress-zone=CUCM

2412.16 SipPacket Call-ID: 1df72600-f711e965-e809e-b02020a@10.2.2.11

2412.16 SipPacket CSeq: 101 INVITE

2412.16 SipPacket Contact:

<217886>

2412.16 SipPacket From: "Tony D'Astolfo"

<216886>;tag=2628263~672ae68a-0b27-4f6e-b42e-35e9e0bc74c3-54922102

2412.16 SipPacket To: "EX90-Test2"

<217886>;tag=cb62d3081f7a2be0

2412.16 SipPacket Record-Route:

2412.16 SipPacket Server: TANDBERG/516 (TC5.1.0.280662)

2412.16 SipPacket Content-Length: 0

2412.16 SipPacket

2412.16 SipPacket >!!!

2412.24 MediaStreamController I: SC::PlayReq(og=14)

path='/sounds/ice_crystals.mp4', tone=0

2412.33 MediaStreamController I: MV::getVCSetting getOutputPortStatus

initialized 1

2412.33 MediaStreamController I: MV::getVCSetting localHwCookieHint_ 0 w

1280 h 720

2412.34 MediaStreamController I: MV::getVCSetting getOutputPortStatus

initialized 0

2412.34 MediaStreamController I: MV::getVCSetting localHwCookieHint_ 32 w

704 h 576

2415.32 IXUser I: iXController(0x12580b98) registerProtocolUser: proto=1,

user=0x125196a0

2415.33 IXUser I: iXController(0x12580b98) registerProtocolUser: proto=2,

user=0x12518388

2415.33 MC I: RemoteParticipant::setRRRate(p=11): Received new remote rate

from protocol: 0 (old: -1)

2415.36 MC I: Conference::addRemoteParticipant(p=11,ch=3) adding

modecontrol shared channel

2415.36 MC I: Conference::modeChanged(c=7) holder=-1 Video (3): vid-off

0x0@0.0 0k

2415.36 MC I: RemoteParticipant::outgoingDuoModeChanged(p=11, ch=3):

ModesController wants to run mode: Video (3): vid-off 0x0@0.0 0k

2415.36 MC !ER ComasListener::getRemoteParticipant(p=?) no appId

2415.36 SipCall I: ==== signalOutgoingModeChange appId=-1, stackId=0,

eventCookie=-1, isEmptyCap=0, holdState=0

2415.36 MC !ER ComasListener::getRemoteParticipant(p=?) no appId

2415.37 MC I: ComasListener::openSendChannel(p=-1): No RemoteParticipant

2415.37 CAMERA I: CamVisca::sendCAMPositionInd cameraId=1 pos=0/0/566/244

2415.37 RTP I: SockPair::dynamic(sock=0) retrying next pair (rtp

port=2386)

2415.37 RTP I: SockPair::dynamic(sock=1) retrying next pair (rtp

port=2388)

2415.37 RTP I: SockPair::dynamic(sock=2) retrying next pair (rtp

port=2390)

2415.37 RTP I: SockPair::dynamic(sock=3) retrying next pair (rtp

port=2392)

2415.38 RTP I: SockPair::dynamic(sock=4) retrying next pair (rtp

port=2394)

2415.38 DATACTRL I: DataGateAddReq(ig=7) assoc gate -1 hdlc=yes

2415.49 RTP I: TrafficCtrl: Added new entry (main): id: 62, tcid: 65568

2415.50 MC I: MediaServerAction(): InputGate(ig=111) creation took 141

miliseconds

2415.50 MC I: MediaServerAction(): InputGate(ig=119) creation took 143

miliseconds

2415.50 MC I: MediaServerAction(): InputGate(ig=114) creation took 147

miliseconds

2415.51 RTP I: TrafficCtrl: Added new entry (duo): id: 64, tcid: 65568

2415.52 MC I: MediaServerAction(): InputGate(ig=117) creation took 166

miliseconds

2415.53 RTP I: TrafficCtrl: Added new entry (duo): id: 68, tcid: 65569

2415.54 MC I: setCallState(p=11) InCall

2415.54 SipCall I: ==== acceptIncomingCall appId=-1, stackId=0,

eventCookie=-1

2415.54 SipMedia W: SIPMEDIA_connectBFCP The BFCP medialine is not

negotiated

2415.55 MC I: RemoteParticipant::modeChanged(p=11, ch=1): ModesController

wants to run mode: Audio (1): aud-off 0k

2415.55 SipCall I: ==== signalOutgoingModeChange appId=11, stackId=0,

eventCookie=-1, isEmptyCap=0, holdState=0

2415.55 MC I: configureOutgoingChannel(p=11, ch=1): Configuring mode :

Audio (1): aud-off 0k (modescontroller says real is: Audio (1): aud-off

0k )

2415.55 MC I: RemoteParticipant::modeChanged(p=11, ch=2): ModesController

wants to run mode: Video (2): vid-off 0x0@0.0 0k

2415.55 SipCall I: ==== signalOutgoingModeChange appId=11, stackId=0,

eventCookie=-1, isEmptyCap=0, holdState=0

2415.56 MC I: configureOutgoingChannel(p=11, ch=2): Configuring mode :

Video (2): vid-off 0x0@0.0 0k (modescontroller says real is: Video (2):

vid-off 0x0@0.0 0k )

2415.56 MC I: Delaying setting of media mode Video (2): vid-off 0x0@0.0 0k

because the RTP addresses are missing.

2415.56 MC I: RemoteParticipant::modeChanged(p=11, ch=3): ModesController

wants to run mode: Video (3): vid-off 0x0@0.0 0k

2415.56 MC I: Conference::calcCommonRate(c=7,ch=3) rate=0

2415.56 MC I: RemoteParticipant::modeChanged(p=11, ch=4): ModesController

wants to run mode: Data (4): data-off 0k

2415.56 SipCall I: ==== signalOutgoingModeChange appId=11, stackId=0,

eventCookie=-1, isEmptyCap=0, holdState=0

2415.56 MC I: configureOutgoingChannel(p=11, ch=4): Configuring mode :

Data (4): data-off 0k (modescontroller says real is: Data (4): data-off

0k )

2415.56 MC I: RemoteParticipant::modeChanged(p=11, ch=5): ModesController

wants to run mode: Video (5): vid-off 0x0@0.0 0k

2415.57 SipCall I: ==== signalOutgoingModeChange appId=11, stackId=0,

eventCookie=-1, isEmptyCap=0, holdState=0

2415.57 MC I: configureOutgoingChannel(p=11, ch=5): Configuring mode :

Video (5): vid-off 0x0@0.0 0k (modescontroller says real is: Video (5):

vid-off 0x0@0.0 0k )

2415.57 MC I: Delaying setting of media mode Video (5): vid-off 0x0@0.0 0k

because the RTP addresses are missing.

2415.57 MC I: RemoteParticipant::proposeMode(p=11, ch=1): Did not find

mixer ((nil)) or channel (0x1251388c)

2415.57 MC I: RemoteParticipant::propose60fpsThreshold(p=11, ch=1): Did

not find mixer ((nil)) or channel (0x1251388c)

2415.57 MC I: RemoteParticipant::reevalRefMode(p=11,ch=1) set ref [Audio

(1): G.711a 64k ] q= auto, t60=6000

2415.57 MC I: RemoteParticipant::reevalRefMode(p=11,ch=2) set ref [Video

(2): vid-off 0x0@0.0 0k ] q= auto, t60=6000

2415.57 MC I: RemoteParticipant::reevalRefMode(p=11,ch=3) set ref [Video

(3): vid-off 0x0@0.0 0k ] q= auto, t60=6000

2415.57 MC I: RemoteParticipant::proposeMode(p=11, ch=4): Did not find

mixer ((nil)) or channel (0x1255854c)

2415.57 MC I: RemoteParticipant::propose60fpsThreshold(p=11, ch=4): Did

not find mixer ((nil)) or channel (0x1255854c)

2415.57 MC I: RemoteParticipant::reevalRefMode(p=11,ch=4) set ref [Data

(4): H.224-HDLC 64k ] q= auto, t60=6000

2415.57 MC I: RemoteParticipant::reevalRefMode(p=11,ch=5) set ref [Video

(5): vid-off 0x0@0.0 0k ] q= auto, t60=6000

2415.57 MC I: MediaServerAction(): InputGate(ig=121) creation took 215

miliseconds

2415.57 SipCall I: sip_call_handler::handleSIPMRtpLocalCapsetInd(11/0/-1,

capsetentries: 27): Received local capset with payload types set

2415.58 MC I: RemoteParticipant::configureIncomingChannel: Capability

table:

1: (2) H.264 maxbr=6000000 maxfs=9000 maxmbps=245000 maxsmbps=245000

maxfps=6000

2: (2) H.264NIL maxbr=6000000 maxfs=9000 maxmbps=245000 maxsmbps=245000

maxfps=6000

3: (2) H.263PP maxbr=2000000 fps-qcif=3000 fps-cif=3000 fps-4cif=3000

fps-sif=3000 fps-4sif=3000 fps-vga=3000 fps-svga=3000 fps-xga=3000

fps-wcif=3000 fps-w4cif=1500 fps-w720=1000 fps-wxga=1000

4: (2) H.263 maxbr=2000000 fps-qcif=3000 fps-cif=3000 fps-4cif=3000

5: (2) H.261 maxbr=2000000 fps-qcif=3000 fps-cif=3000

6: (3) H.264 maxbr=6000000 maxfs=9000 maxmbps=245000 maxsmbps=245000

maxfps=6000 feat=Pres-Token

7: (3) H.264NIL maxbr=6000000 maxfs=9000 maxmbps=245000 maxsmbps=245000

maxfps=6000 feat=Pres-Token

8: (3) H.263PP maxbr=2000000 fps-qcif=3000 fps-cif=3000 fps-4cif=3000

fps-sif=3000 fps-4sif=3000 fps-vga=3000 fps-svga=3000 fps-xga=3000

fps-wcif=3000 fps-w4cif=1500 fps-w720=1000 fps-wxga=1000 feat=Pres-Token

9: (3) H.263 maxbr=2000000 fps-qcif=3000 fps-cif=3000 fps-4cif=3000

feat=Pres-Token

10: (3) H.261 maxbr=2000000 fps-qcif=3000 fps-cif=3000 feat=Pres-Token

11: (1) AAC-LD rate=128k profile=25

12: (1) AAC-LD rate=64k profile=24

13: (1) AAC-LD rate=56k profile=24

14: (1) AAC-LD rate=48k profile=24

15: (1) G.722.1 rate=32k ptime(ms)=40

16: (1) G.722.1 rate=24k ptime(ms)=40

17: (1) G.722 rate=64k ptime(ms)=40

18: (1) G.711a rate=64k ptime(ms)=40

19: (1) G.711mu rate=64k ptime(ms)=40

20: (1) RTP-Dtmf

21: (2) RTCP-Feedback-PLI

21: (2) RTCP-Feedback-FIR

21: (3) RTCP-Feedback-PLI

21: (3) RTCP-Feedback-FIR

22: (3) H.239Ctrl

23: (4) H.224-HDLC

23: (4) H.224

Simcapset:

(1) {11,12,13,14,15,16,17,18,19,20}{1,2,3,4,5}{6,7,8,9,10}{23,23}

2415.58 SipCall I: sip_call_handler::SIPM_Crypto_Ind (11/0/0)

encryptionActive=0

2415.60 VIDEOCTRL-0 I: (proxy,161) maximum number of framestorage slices 2

2415.60 DATACTRL I: DataGateCfgReq(ig=7) hdlc=yes

2415.61 SipStack I: SipUa GRUU added OK

2415.61 SipPacket PacketDump: Proto: SIP, Name: SIP/2.0 200 OK,

Direction: Outgoing, RemoteAddress: 10.2.2.27:5061, GroupEntity:

1df72600-f711e965-e809e-b02020a@10.2.2.11, Time: 2415610, Content: !!!<

2415.61 SipPacket SIP/2.0 200 OK

2415.61 SipPacket Via: SIP/2.0/TLS

10.2.2.27:5061;egress-zone=DefaultSubZone;branch=z9hG4bK53462630b7e5546a2c7fbf752eed6e6c1467.8830a1b6a042c47e99b7857db0440f2d;proxy-call-id=1f872884-7829-11e1-b44a-0010f31f232e;received=10.2.2.27;rport=5061

2415.61 SipPacket Via: SIP/2.0/TCP

10.2.2.11:5060;branch=z9hG4bK11bd4d4ccd339d;received=10.2.2.11;ingress-zone=CUCM

2415.61 SipPacket Call-ID: 1df72600-f711e965-e809e-b02020a@10.2.2.11

2415.61 SipPacket CSeq: 101 INVITE

2415.62 SipPacket Contact:

<217886>

2415.62 SipPacket From: "Tony D'Astolfo"

<216886>;tag=2628263~672ae68a-0b27-4f6e-b42e-35e9e0bc74c3-54922102

2415.62 SipPacket To: "EX90-Test2"

<217886>;tag=cb62d3081f7a2be0

2415.62 SipPacket Record-Route:

2415.62 SipPacket Allow:

INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY

2415.62 SipPacket Server: TANDBERG/516 (TC5.1.0.280662)

2415.62 SipPacket Supported: replaces,100rel,timer,gruu,path,outbound

2415.62 SipPacket Require: timer

2415.62 SipPacket Session-Expires: 1800;refresher=uas

2415.62 SipPacket Min-SE: 1800

2415.62 SipPacket Content-Type: application/sdp

2415.62 SipPacket Content-Length: 2495

2415.62 SipPacket

2415.62 SipPacket v=0

2415.62 SipPacket o=tandberg 11 1 IN IP4 10.2.15.177

2415.62 SipPacket s=-

2415.62 SipPacket c=IN IP4 10.2.15.177

2415.62 SipPacket b=AS:6000

2415.62 SipPacket t=0 0

2415.62 SipPacket m=audio 2386 RTP/AVP 100 102 103 104 105 106 9 8 0 101

2415.62 SipPacket b=TIAS:128000

2415.62 SipPacket a=rtpmap:100 MP4A-LATM/90000

2415.62 SipPacket a=fmtp:100

profile-level-id=25;object=23;bitrate=128000

2415.62 SipPacket a=rtpmap:102 MP4A-LATM/90000

2415.62 SipPacket a=fmtp:102 profile-level-id=24;object=23;bitrate=64000

2415.62 SipPacket a=rtpmap:103 MP4A-LATM/90000

2415.62 SipPacket a=fmtp:103 profile-level-id=24;object=23;bitrate=56000

2415.62 SipPacket a=rtpmap:104 MP4A-LATM/90000

2415.62 SipPacket a=fmtp:104 profile-level-id=24;object=23;bitrate=48000

2415.62 SipPacket a=rtpmap:105 G7221/16000

2415.62 SipPacket a=fmtp:105 bitrate=32000

2415.62 SipPacket a=rtpmap:106 G7221/16000

2415.62 SipPacket a=fmtp:106 bitrate=24000

2415.63 SipPacket a=rtpmap:9 G722/8000

2415.63 SipPacket a=rtpmap:8 PCMA/8000

2415.63 SipPacket a=rtpmap:0 PCMU/8000

2415.63 SipPacket a=rtpmap:101 telephone-event/8000

2415.63 SipPacket a=fmtp:101 0-15

2415.63 SipPacket a=sendrecv

2415.63 SipPacket m=video 2388 RTP/AVP 97 98 99 34 31

2415.63 SipPacket b=TIAS:6000000

2415.63 SipPacket a=rtpmap:97 H264/90000

2415.63 SipPacket a=fmtp:97

profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000

2415.63 SipPacket a=rtpmap:98 H264/90000

2415.63 SipPacket a=fmtp:98

profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;packetization-mode=1;max-fps=6000

2415.63 SipPacket a=rtpmap:99 H263-1998/90000

2415.63 SipPacket a=fmtp:99

custom=1280,768,3;custom=1280,720,3;custom=1024,768,1;custom=1024,576,2;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20000

2415.64 SipPacket a=rtpmap:34 H263/90000

2415.64 SipPacket a=fmtp:34 cif4=1;cif=1;qcif=1;maxbr=20000

2415.64 SipPacket a=rtpmap:31 H261/90000

2415.64 SipPacket a=fmtp:31 cif=1;qcif=1;maxbr=20000

2415.64 SipPacket a=rtcp-fb:* nack pli

2415.64 SipPacket a=rtcp-fb:* ccm fir

2415.64 SipPacket a=sendrecv

2415.64 SipPacket a=content:main

2415.64 SipPacket a=label:11

2415.64 SipPacket a=answer:full

2415.64 SipPacket m=application 5070 UDP/BFCP *

2415.64 SipPacket a=floorctrl:c-s

2415.64 SipPacket a=confid:1

2415.64 SipPacket a=floorid:2 mstrm:12

2415.64 SipPacket a=userid:11

2415.64 SipPacket a=setup:actpass

2415.64 SipPacket a=connection:new

2415.64 SipPacket m=video 2390 RTP/AVP 97 98 99 34 31

2415.64 SipPacket b=TIAS:6000000

2415.64 SipPacket a=rtpmap:97 H264/90000

2415.64 SipPacket a=fmtp:97

profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000

2415.64 SipPacket a=rtpmap:98 H264/90000

2415.64 SipPacket a=fmtp:98

profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;packetization-mode=1;max-fps=6000

2415.64 SipPacket a=rtpmap:99 H263-1998/90000

2415.64 SipPacket a=fmtp:99

custom=1280,768,3;custom=1280,720,3;custom=1024,768,1;custom=1024,576,2;custom=800,600,1;cif4=1;custom=720,480,1;custom=640,480,1;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=20000

2415.64 SipPacket a=rtpmap:34 H263/90000

2415.64 SipPacket a=fmtp:34 cif4=1;cif=1;qcif=1;maxbr=20000

2415.65 SipPacket a=rtpmap:31 H261/90000

2415.65 SipPacket a=fmtp:31 cif=1;qcif=1;maxbr=20000

2415.65 SipPacket a=rtcp-fb:* nack pli

2415.65 SipPacket a=rtcp-fb:* ccm fir

2415.65 SipPacket a=sendrecv

2415.65 SipPacket a=content:slides

2415.65 SipPacket a=label:12

2415.65 SipPacket m=application 2392 RTP/AVP 107

2415.65 SipPacket a=rtpmap:107 H224/4800

2415.65 SipPacket a=sendrecv

2415.65 SipPacket >!!!

2415.67 SipPacket PacketDump: Proto: SIP, Name: ACK

sip:217886@10.2.15.177:5061;transport=tls SIP/2.0, Direction: Incoming,

RemoteAddress: 10.2.2.27:5061, GroupEntity:

1df72600-f711e965-e809e-b02020a@10.2.2.11, Time: 2415670, Content: !!!<

2415.67 SipPacket ACK sip:217886@10.2.15.177:5061;transport=tls SIP/2.0

2415.67 SipPacket Via: SIP/2.0/TLS

10.2.2.27:5061;egress-zone=DefaultSubZone;branch=z9hG4bKe6f7d96e3a96baf44c3f4410c2a226081470.8830a1b6a042c47e99b7857db0440f2d;proxy-call-id=1f872884-7829-11e1-b44a-0010f31f232e;rport

2415.67 SipPacket Via: SIP/2.0/TCP

10.2.2.11:5060;branch=z9hG4bK11bd4f6d524810;received=10.2.2.11;ingress-zone=CUCM

2415.67 SipPacket Call-ID: 1df72600-f711e965-e809e-b02020a@10.2.2.11

2415.67 SipPacket CSeq: 101 ACK

2415.67 SipPacket From: "Tony D'Astolfo"

<216886>;tag=2628263~672ae68a-0b27-4f6e-b42e-35e9e0bc74c3-54922102

2415.67 SipPacket To: "EX90-Test2"

<217886>;tag=cb62d3081f7a2be0

2415.67 SipPacket Max-Forwards: 69

2415.67 SipPacket Date: Tue, 27 Mar 2012 16:23:01 GMT

2415.67 SipPacket Allow-Events: presence,kpml

2415.67 SipPacket X-TAATag: 1f872960-7829-11e1-becd-0010f31f232e

2415.67 SipPacket Content-Length: 0

2415.67 SipPacket

2415.67 SipPacket >!!!

2415.68 SipStack !ER SIP_ASSERT, File:

functional/protocols/sipmedia/SipMedia.cpp, Line 2857

2415.68 SipCall I: sip_call_handler::SIPM_Call_Res_Ack (11/0/-1)

2415.69 MC W: Mcfsm::handleOtherMessages() unknown msg

SIPM_Internal_Exception from EPAppl

2415.69 MC I: VideoInputGateImpl::inputGateConfigState(ig=114):

DecoderConfig[114]: maxMbps: 244980 maxWidth: 1920 maxHeight: 1200

memsize: 74432424

2415.70 SipPacket PacketDump: Proto: SIP, Name: BYE

sip:217886@10.2.15.177:5061;transport=tls SIP/2.0, Direction: Incoming,

RemoteAddress: 10.2.2.27:5061, GroupEntity:

1df72600-f711e965-e809e-b02020a@10.2.2.11, Time: 2415703, Content: !!!<

2415.70 SipPacket BYE sip:217886@10.2.15.177:5061;transport=tls SIP/2.0

2415.71 SipPacket Via: SIP/2.0/TLS

10.2.2.27:5061;egress-zone=DefaultSubZone;branch=z9hG4bK832a7a38be76b71de6eb2cef3222b2bb1471.8830a1b6a042c47e99b7857db0440f2d;proxy-call-id=1f872884-7829-11e1-b44a-0010f31f232e;rport

2415.71 SipPacket Via: SIP/2.0/TCP

10.2.2.11:5060;branch=z9hG4bK11bd5027e4920f;received=10.2.2.11;ingress-zone=CUCM

2415.71 SipPacket Call-ID: 1df72600-f711e965-e809e-b02020a@10.2.2.11

2415.71 SipPacket CSeq: 102 BYE

2415.71 SipPacket From: "Tony D'Astolfo"

<216886>;tag=2628263~672ae68a-0b27-4f6e-b42e-35e9e0bc74c3-54922102

2415.71 SipPacket To: "EX90-Test2"

<217886>;tag=cb62d3081f7a2be0

2415.71 SipPacket Max-Forwards: 69

2415.71 SipPacket User-Agent: Cisco-CUCM8.5

2415.71 SipPacket Date: Tue, 27 Mar 2012 16:23:01 GMT

2415.71 SipPacket X-TAATag: 1f872960-7829-11e1-becd-0010f31f232e

2415.71 SipPacket Content-Length: 0

2415.71 SipPacket

2415.71 SipPacket >!!!

2415.71 SipStack I: SipUa GRUU added OK

2415.72 SipPacket PacketDump: Proto: SIP, Name: SIP/2.0 200 OK,

Direction: Outgoing, RemoteAddress: 10.2.2.27:5061, GroupEntity:

1df72600-f711e965-e809e-b02020a@10.2.2.11, Time: 2415715, Content: !!!<

2415.72 SipPacket SIP/2.0 200 OK

2415.72 SipPacket Via: SIP/2.0/TLS

10.2.2.27:5061;egress-zone=DefaultSubZone;branch=z9hG4bK832a7a38be76b71de6eb2cef3222b2bb1471.8830a1b6a042c47e99b7857db0440f2d;proxy-call-id=1f872884-7829-11e1-b44a-0010f31f232e;received=10.2.2.27;rport=5061

2415.72 SipPacket Via: SIP/2.0/TCP

10.2.2.11:5060;branch=z9hG4bK11bd5027e4920f;received=10.2.2.11;ingress-zone=CUCM

2415.72 SipPacket Call-ID: 1df72600-f711e965-e809e-b02020a@10.2.2.11

2415.72 SipPacket CSeq: 102 BYE

2415.72 SipPacket Contact:

<217886>

2415.72 SipPacket From: "Tony D'Astolfo"

<216886>;tag=2628263~672ae68a-0b27-4f6e-b42e-35e9e0bc74c3-54922102

2415.72 SipPacket To: "EX90-Test2"

<217886>;tag=cb62d3081f7a2be0

2415.72 SipPacket Allow:

INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY

2415.72 SipPacket Server: TANDBERG/516 (TC5.1.0.280662)

2415.72 SipPacket Supported: replaces,100rel,timer,gruu,path,outbound

2415.72 SipPacket Content-Length: 0

2415.72 SipPacket

2415.72 SipPacket >!!!

2415.72 SipCall I: sip_call_handler::handleSIPMDiscInd(11/0/0): Incoming

disconnect (reason: , sipCause.dc=8, sipCause.status=0)

2415.73 MC I: RemoteParticipant::reevalRefMode(p=11,ch=2) set ref [Video

(2): vid-off 0x0@0.0 0k ] q= auto, t60=6000

2415.73 MC I: RemoteParticipant::reevalRefMode(p=11,ch=5) set ref [Video

(5): vid-off 0x0@0.0 0k ] q= auto, t60=6000

2415.76 SipCall I: ==== affirmIncomingDisconnect appId=11, stackId=0,

eventCookie=-1

2415.80 IXUser I: iXController teardownIxChannel: Not connected

2415.80 MC I: CapabilityControllerImpl::setCapset() reduced = 0,

waitForDuoGate = 0, hasLegacyVideo = 0

2415.82 AUDIOCTRL-0 W: rem lig 1 from mix 8 failed: gate not added to mix

2415.83 RTP I: TrafficCtrl: Deleting entry (main): id: 62, tcid: 65568

2415.83 RTP I: TrafficCtrl: Deleting entry (duo): id: 64, tcid: 65568

2415.84 RTP I: TrafficCtrl: Deleting entry (duo): id: 68, tcid: 65569

2415.84 CAMERA I: CamVisca::Ready_doCAMActionReq cameraId=1 actionId=20

2415.84 MediaStreamController I: MV::getVCSetting getOutputPortStatus

initialized 1

2415.84 MediaStreamController I: MV::getVCSetting localHwCookieHint_ 0 w

1280 h 720

2415.85 MediaStreamController I: MV::getVCSetting getOutputPortStatus

initialized 1

2415.85 MediaStreamController I: MV::getVCSetting localHwCookieHint_ 24 w

1920 h 1200

2415.85 MediaStreamController I: MV::getVCSetting getOutputPortStatus

initialized 0

2415.85 MediaStreamController I: MV::getVCSetting localHwCookieHint_ 32 w

704 h 576

2415.86 MediaStreamController I: doAUDIOMIXERREMGATECNF(ms-ig=9) not found

2415.86 MediaStreamController !ER doAUDIOMIXERREMGATEREJ(ms-mx=8) not

found

2415.86 DATACTRL I: DataGateRemReq(ig=7)

2415.86 MC W: Mcfsm::handleOtherMessages() unknown msg SIP_DialogFreed_Ind

from SipStack

2415.87 VIDEOCTRL-0 I: VIDEOCTRL_doReadyMixerConfigureReq: redundant

signal for mixer (transcoding,16) from (MSCtrl,0)

2415.87 VIDEOCTRL-0 I: VIDEOCTRL_doReadyMixerConfigureReq: redundant

signal for mixer (transcoding,17) from (MSCtrl,0)

2415.88 MediaStreamController !ER doVIDEOGATECONFIGURECNF(ms-ig=29) not

found

2415.89 MediaStreamController !ER doVIDEOMIXERCONFIGURECNF(ms-mx=16) not

found

2415.89 MediaStreamController !ER doVIDEOMIXERCONFIGURECNF(ms-mx=17) not

found

2416.00 MediaStreamController I: MV::getVCSetting getOutputPortStatus

initialized 1

2416.00 MediaStreamController I: MV::getVCSetting localHwCookieHint_ 0 w

1280 h 720

2416.00 MediaStreamController I: MV::getVCSetting getOutputPortStatus

initialized 0

2416.00 MediaStreamController I: MV::getVCSetting localHwCookieHint_ 32 w

704 h 576

2416.45 VIDEOCTRL-0 I: (proxy,161) maximum number of framestorage slices 2

2416.50 VIDEOCTRL-0 I: VIDEOCTRL_doReadyMixerConfigureReq: redundant

signal for mixer (rawvideo,1) from (MSCtrl,0)

2416.50 VIDEOCTRL-0 I: VIDEOCTRL_doReadyMixerConfigureReq: redundant

signal for mixer (rawvideo,2) from (MSCtrl,0)

2416.50 VIDEOCTRL-0 I: VIDEOCTRL_doReadyMixerConfigureReq: redundant

signal for mixer (rawvideo,3) from (MSCtrl,0)

2416.51 MC I: MediaServerAction(): InputGate(ig=124) creation took 704

miliseconds

2426.40 CAMERA I: CamVisca::sendCAMPositionInd cameraId=1

pos=34521644/35329252/566/244

xstatus

*s Audio Microphones Mute: Off

*s Audio Volume: 20

*s Audio SelectedOutputConnector: Internal

*s Camera 1 Connected: True

*s Camera 1 HardwareID: ""

*s Camera 1 Manufacturer: "Cisco"

*s Camera 1 Model: "EX90"

*s Camera 1 SoftwareID: ""

*s Camera 1 SerialNumber: "XXXXXXX"

*s Camera 1 IpAddress: ""

*s Camera 1 MacAddress: ""

*s Camera 1 Position Pan: 0

*s Camera 1 Position Tilt: 0

*s Camera 1 Position Zoom: 566

*s Camera 1 Position Focus: 244

*s Camera 1 Capabilities Options: "zf"

*s Camera 1 Flip: "Off"

*s Conference Presentation Mode: Off

*s Conference Presentation Protocol: ""

*s Conference Presentation Resolution Height: 0

*s Conference Presentation Resolution Width: 0

*s Conference Presentation SiteId: 0

*s Conference Presentation LocalSource: 0

*s Conference Multipoint Mode: "MultiSite"

*s Conference DoNotDisturb: Inactive

*s Experimental Audio Input Connectors Extmic 1 ConnectionStatus:

NotConnected

*s Experimental Audio Output Connectors Headphone 1 ConnectionStatus:

NotConnected

*s H323 Gatekeeper Status: Registered

*s H323 Gatekeeper Address: "10.2.2.27"

*s H323 Gatekeeper Port: 1719

*s H323 Gatekeeper Reason: ""

*s HttpFeedback 1 URL: ""

*s HttpFeedback 1 Expression: ""

*s HttpFeedback 2 URL: ""

*s HttpFeedback 2 Expression: ""

*s HttpFeedback 3 URL: "

http://tms.sncorp.com/tms/public/feedback/code.aspx"

*s HttpFeedback 3 Expression: "/History/CallLog/History"

*s HttpFeedback 3 Expression: "/History/CallLogs/Call"

*s HttpFeedback 3 Expression: "/Status/Call[Status='Connected']"

*s HttpFeedback 3 Expression: "/Status/H323/Gatekeeper"

*s HttpFeedback 3 Expression: "/Status/SIP/Registration"

*s HttpFeedback 3 Expression: "/Status/Network/Ethernet/Speed"

*s HttpFeedback 3 Expression: "/Event/CallSuccessful"

*s HttpFeedback 3 Expression: "/Event/Message/Prompt/Response"

*s HttpFeedback 3 Expression: ""

*s HttpFeedback 4 URL: ""

*s HttpFeedback 4 Expression: ""

*s Network 1 Ethernet MacAddress: "00:50:60:06:78:75"

*s Network 1 Ethernet Speed: "100full"

*s Network 1 IPv4 Address: "10.2.15.177"

*s Network 1 IPv4 SubnetMask: "255.255.255.0"

*s Network 1 IPv4 Gateway: "10.2.15.1"

*s Network 1 IPv4 DNS Domain Name: "sncorp.intranet.com"

*s Network 1 IPv4 DNS Server 1 Address: "10.2.40.200"

*s Network 1 IPv4 DNS Server 2 Address: "10.2.40.201"

*s Network 1 IPv4 DNS Server 3 Address: ""

*s Network 1 IPv4 DNS Server 4 Address: ""

*s Network 1 IPv4 DNS Server 5 Address: ""

*s Network 1 IPv6 Address: ""

*s Network 1 IPv6 Gateway: ""

*s Network 1 MTU: 1500

*s Network 1 VLAN Voice VlanId: "Off"

*s Preset 1 Defined: False

*s Preset 1 Type: All

*s Preset 1 Description: ""

*s Preset 2 Defined: False

*s Preset 2 Type: All

*s Preset 2 Description: ""

*s Preset 3 Defined: False

*s Preset 3 Type: All

*s Preset 3 Description: ""

*s Preset 4 Defined: False

*s Preset 4 Type: All

*s Preset 4 Description: ""

*s Preset 5 Defined: False

*s Preset 5 Type: All

*s Preset 5 Description: ""

*s Preset 6 Defined: False

*s Preset 6 Type: All

*s Preset 6 Description: ""

*s Preset 7 Defined: False

*s Preset 7 Type: All

*s Preset 7 Description: ""

*s Preset 8 Defined: False

*s Preset 8 Type: All

*s Preset 8 Description: ""

*s Preset 9 Defined: False

*s Preset 9 Type: All

*s Preset 9 Description: ""

*s Preset 10 Defined: False

*s Preset 10 Type: All

*s Preset 10 Description: ""

*s Preset 11 Defined: False

*s Preset 11 Type: All

*s Preset 11 Description: ""

*s Preset 12 Defined: False

*s Preset 12 Type: All

*s Preset 12 Description: ""

*s Preset 13 Defined: False

*s Preset 13 Type: All

*s Preset 13 Description: ""

*s Preset 14 Defined: False

*s Preset 14 Type: All

*s Preset 14 Description: ""

*s Preset 15 Defined: False

*s Preset 15 Type: All

*s Preset 15 Description: ""

*s Provisioning Status: Provisioned

*s Provisioning Reason: ""

*s Provisioning Software UpgradeStatus SessionId: ""

*s Provisioning Software UpgradeStatus LastChange: ""

*s Provisioning Software UpgradeStatus Status:

*s Provisioning Software UpgradeStatus Phase:

*s Provisioning Software UpgradeStatus Message: ""

*s Provisioning Software UpgradeStatus VersionId: ""

*s Provisioning Software UpgradeStatus URL: ""

*s Provisioning Software Current VersionId: ""

*s Provisioning Software Current URL: ""

*s Provisioning Software Current CompletedAt: ""

*s Security FIPS Mode: Off

*s SIP Proxy 1 Status: Active

*s SIP Proxy 1 Address: "10.2.2.27"

*s SIP Proxy 1 Secure: True

*s SIP Proxy 1 Verified: False

*s SIP Proxy 2 Status: Unknown

*s SIP Proxy 2 Address: "10.2.2.10"

*s SIP Proxy 2 Secure: False

*s SIP Proxy 2 Verified: False

*s SIP Registration 1 Status: Registered

*s SIP Registration 1 Reason: ""

*s SIP Registration 1 URI: "217886@sncorp.com"

*s SIP Registration 1 Authentication: Off

*s SIP Registration 2 Status: Failed

*s SIP Registration 2 Reason: "Failed to send"

*s SIP Registration 2 URI: "217886@sncorp.com"

*s SIP Profile 1 Proxy 1 Status: Active

*s SIP Profile 1 Proxy 1 Address: "10.2.2.27"

*s SIP Profile 1 Proxy 2 Status: Unknown

*s SIP Profile 1 Proxy 2 Address: "10.2.2.10"

*s SIP Profile 1 Secure: False

*s SIP Profile 1 Verified: False

*s SIP Profile 1 Authentication: Off

*s SIP Profile 1 Mailbox MessagesWaiting: 0

*s SIP Profile 1 Registration 1 Status: Registered

*s SIP Profile 1 Registration 1 Reason: ""

*s SIP Profile 1 Registration 1 URI: "217886@sncorp.com"

*s SIP Profile 1 Registration 2 Status: Failed

*s SIP Profile 1 Registration 2 Reason: "Failed to send"

*s SIP Profile 1 Registration 2 URI: "217886@sncorp.com"

*s Standby Active: Off

*s SystemUnit ProductType: "TANDBERG Codec"

*s SystemUnit ProductId: "TANDBERG EX90"

*s SystemUnit ProductPlatform: "EX90"

*s SystemUnit Uptime: 2465

*s SystemUnit Software Application: "Endpoint"

*s SystemUnit Software Version: "TC5.1.0.280662"

*s SystemUnit Software Name: "s52000"

*s SystemUnit Software ReleaseDate: "2012-02-14"

*s SystemUnit Software MaxVideoCalls: 3

*s SystemUnit Software MaxAudioCalls: 4

*s SystemUnit Software ReleaseKey: "true"

*s SystemUnit Software OptionKeys MultiSite: "true"

*s SystemUnit Software OptionKeys PremiumResolution: "true"

*s SystemUnit Software OptionKeys DualDisplay: "true"

*s SystemUnit Hardware Module SerialNumber: "A1AR04E00388"

*s SystemUnit Hardware Module Identifier: "05"

*s SystemUnit Hardware Module CompatibilityLevel: "0"

*s SystemUnit Hardware MainBoard SerialNumber: "F1082762"

*s SystemUnit Hardware MainBoard Identifier: "101540-8 "

*s SystemUnit Hardware BootSoftware: "U-Boot 2010.06-94"

*s SystemUnit Hardware MonitoringSoftware: "43"

*s SystemUnit Hardware Monitoring Fan 1 Status: "locked on 1200 rpm"

*s SystemUnit Hardware Monitoring Fan 2 Status: "not present"

*s SystemUnit Hardware Temperature: "60.5"

*s SystemUnit State System: Initialized

*s SystemUnit State MaxNumberOfCalls: 4

*s SystemUnit State MaxNumberOfActiveCalls: 4

*s SystemUnit State NumberOfActiveCalls: 0

*s SystemUnit State NumberOfSuspendedCalls: 0

*s SystemUnit State NumberOfInProgressCalls: 0

*s SystemUnit State Subsystem Application: Initialized

*s SystemUnit State CameraLid: Closed

*s SystemUnit State CameraTilt: Off

*s SystemUnit State Cradle: Down

*s SystemUnit ContactInfo: "217886@sncorp.com"

*s Time ZoneOlson: America/Los_Angeles

*s Video Input Source 1 Resolution Height: 0

*s Video Input Source 1 Resolution Width: 0

*s Video Input Source 1 Resolution RefreshRate: 0

*s Video Input Source 1 Resolution FormatType: Unknown

*s Video Input Source 1 Resolution FormatStatus: Error

*s Video Input Source 2 Resolution Height: 0

*s Video Input Source 2 Resolution Width: 0

*s Video Input Source 2 Resolution RefreshRate: 0

*s Video Input Source 2 Resolution FormatType: Unknown

*s Video Input Source 2 Resolution FormatStatus: Error

*s Video Input Source 3 Resolution Height: 1080

*s Video Input Source 3 Resolution Width: 1920

*s Video Input Source 3 Resolution RefreshRate: 60

*s Video Input Source 3 Resolution FormatType: Digital

*s Video Input Source 3 Resolution FormatStatus: Ok

*s Video Input HDMI 1 Connected: False

*s Video Input HDMI 1 SignalState: Unknown

*s Video Input DVI 2 Connected: False

*s Video Input DVI 2 SignalState: Unknown

*s Video Output HDMI 1 Resolution Height: 720

*s Video Output HDMI 1 Resolution Width: 1280

*s Video Output HDMI 1 Resolution RefreshRate: 60

*s Video Output LCD 2 Resolution Height: 1200

*s Video Output LCD 2 Resolution Width: 1920

*s Video Output LCD 2 Resolution RefreshRate: 60

*s Video Layout PresentationView: Default

*s Video Layout Site 1 Output 1 FamilyName: "fullscreen"

*s Video Layout Site 1 Output 1 FullFamilyName:

"fullscreen-local-single-default"

*s Video Layout Site 1 Output 1 GraphicName: "1"

*s Video Layout Site 1 Output 1 Frame 1 PositionX: 0

*s Video Layout Site 1 Output 1 Frame 1 PositionY: 0

*s Video Layout Site 1 Output 1 Frame 1 Width: 10000

*s Video Layout Site 1 Output 1 Frame 1 Height: 10000

*s Video Layout Site 1 Output 1 Frame 1 Layer: 1

*s Video Layout Site 1 Output 1 Frame 1 VideoSourceType: "graphic"

*s Video Layout Site 1 Output 1 Frame 1 VideoSourceId: 0

*s Video Layout Site 1 Output 1 Frame 1 InputNumber: 0

*s Video Layout Site 1 Output 1 Frame 1 Filename:

"/apps/posters/fbg09.png"

*s Video Layout Site 1 Output 3 FamilyName: "fullscreen"

*s Video Layout Site 1 Output 3 FullFamilyName:

"fullscreen-local-single-default"

*s Video Layout Site 1 Output 3 GraphicName: "1"

*s Video Layout Site 1 Output 3 Frame 1 PositionX: 0

*s Video Layout Site 1 Output 3 Frame 1 PositionY: 0

*s Video Layout Site 1 Output 3 Frame 1 Width: 10000

*s Video Layout Site 1 Output 3 Frame 1 Height: 10000

*s Video Layout Site 1 Output 3 Frame 1 Layer: 1

*s Video Layout Site 1 Output 3 Frame 1 VideoSourceType: "graphic"

*s Video Layout Site 1 Output 3 Frame 1 VideoSourceId: 0

*s Video Layout Site 1 Output 3 Frame 1 InputNumber: 0

*s Video Layout Site 1 Output 3 Frame 1 Filename:

"/apps/posters/fbg09.png"

*s Video Layout Local Type: fullscreen

*s Video Layout Local PositionX: 0

*s Video Layout Local PositionY: 0

*s Video Layout Local Width: 0

*s Video Layout Local Height: 0

end

xconfig

*c xConfiguration Audio VolumeHandset: 70

*c xConfiguration Audio VolumeHeadset: 70

*c xConfiguration Audio PreferredOutputConnector: Internal

*c xConfiguration Audio Microphones Mute Enabled: True

*c xConfiguration Audio InternalSpeaker Mode: On

*c xConfiguration Audio Volume: 20

*c xConfiguration Audio SoundsAndAlerts RingVolume: 50

*c xConfiguration Audio SoundsAndAlerts RingTone: IceCrystals

*c xConfiguration Audio SoundsAndAlerts KeyTones Mode: On

*c xConfiguration Cameras PowerLine Frequency: Auto

*c xConfiguration Cameras Camera 1 Brightness Mode: Auto

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end

OK

From: mubakhta

To: anthonymd

Date: 03/27/2012 08:22 AM

Subject: - Re: EX90 and CUCM

Home

Re: EX90 and CUCM

created by Mubashshir Akhtar in TelePresence - View the full discussion

Hi Anthony,

Can you make sure that you are running the log command on the Ex90 in the

API interface?

You should be logged in as admin and then try to run this command on the

Ex90. Also, I had mentioned about the Media Termination Point.

Can you verify the setting of the Media Termination Point on the CUCM in

the phone profiles and on the SIP Trunk?

Regards,

Mubashshir Akhtar

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Start a new discussion in TelePresence at Home

mubakhta · ‎03-26-2012

Hi Anthony,

Please check the Media Termination Point setting from the device profiles of both the PSTN phone and the Ex90 and make sure that the Media Termination Point is not enabled. Also, check it in the case of the SIP Trunk between the CUCM and VCS and make sure Media Termination Point Required is unchecked. In case if it is checked, the call would require a Termination Point and in the absence, it would ultimately fail. Try to disable it and then share the results.

Regards,

Mubashshir Akhtar

Thanks Mubashshir Akhtar

Alok Jaiswal · ‎03-26-2012

Hi Anthony,

So this is my understanding of your setup.

Calls come from PSTN gateway router and hits to CUCM to a DN. This DN as you mentioned is configured for SNR and then CUCM routes this to remote destination configured under SNR for this DN.

The call then reaches to VCS via SIP trunk and hit the Ex90. But when you answer the call it disconnnects.

Now one thing i am sure is that when the call comes from PSTN it would be G.711 codec while it goes to ex90 over the SIP trunk it could be different codec.

Now i have some quesitons.

- What is the DTMF type set to on SIP trunk on CUCM

- what is the CUCM version and VCS software version

- Are you using any MTP on the SIP trunk

- do you have any Hardware trasncoders configured on CUCM

In your case this sounds to be like a codec mismatch and as soon as you answer the call the media is not able to connect due to this mismatch. So i would suggest to use the MTP on SIP trunk and use the appropriate MRGL which should have hardware transcoders included and save the trunk and reset it.

if you are using higher version of CUCM then you can also use the normalization script on this SIP trunk.

After that test the call again.

Thanks

Alok

What you can do is to enable the check box for MTP option required on the SIP trunk configured to VCS.

anthonymd · ‎03-27-2012

Hello Alok,

Please see answers below:

So this is my understanding of your setup.

Calls come from PSTN gateway router and hits to CUCM to a DN. This DN as

you mentioned is configured for SNR and then CUCM routes this to remote

destination configured under SNR for this DN.

The call then reaches to VCS via SIP trunk and hit the Ex90. But when you

answer the call it disconnnects.

The above is correct

Now one thing i am sure is that when the call comes from PSTN it would be

G.711 codec while it goes to ex90 over the SIP trunk it could be different

codec.

Now i have some quesitons.

- What is the DTMF type set to on SIP trunk on CUCM

No Preference

- what is the CUCM version and VCS software version

System version: 8.5.1.11900-21

Version: X7.0.2

- Are you using any MTP on the SIP trunk

No

- do you have any Hardware trasncoders configured on CUCM

Yes

Thanks for your help.

From: aljaiswa

To: anthonymd

Date: 03/26/2012 09:08 PM

Subject: - Re: EX90 and CUCM

Home

Re: EX90 and CUCM

created by Alok Jaiswal in TelePresence - View the full discussion

Hi Anthony,

So this is my understanding of your setup.

Calls come from PSTN gateway router and hits to CUCM to a DN. This DN as

you mentioned is configured for SNR and then CUCM routes this to remote

destination configured under SNR for this DN.

The call then reaches to VCS via SIP trunk and hit the Ex90. But when you

answer the call it disconnnects.

Now one thing i am sure is that when the call comes from PSTN it would be

G.711 codec while it goes to ex90 over the SIP trunk it could be different

codec.

Now i have some quesitons.

- What is the DTMF type set to on SIP trunk on CUCM

- what is the CUCM version and VCS software version

- Are you using any MTP on the SIP trunk

- do you have any Hardware trasncoders configured on CUCM

In your case this sounds to be like a codec mismatch and as soon as you

answer the call the media is not able to connect due to this mismatch. So

i would suggest to use the MTP on SIP trunk and use the appropriate MRGL

which should have hardware transcoders included and save the trunk and

reset it.

if you are using higher version of CUCM then you can also use the

normalization script on this SIP trunk.

After that test the call again.

Thanks

Alok

What you can do is to enable the check box for MTP option required on the

SIP trunk configured to VCS.

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Start a new discussion in TelePresence at Home

Alok Jaiswal · ‎03-27-2012

Anthony,

first thing is to set the DTMF on SIP trunk to RFC2833 save the trunk and reset the trunk and test the calls. However i am not sure will it make much of effect but worth trying.

second if call still disconnects after the above setting try to check the MTP required option on SIP trunk and use the MRGL which has hardware transcoders available in it. Save the trunk and reset the trunk.

After this test the call again.

Thanks

Alok

anthonymd · ‎03-28-2012

Alok,

I tried the above suggestions and still not working. I'll keep poking around but I think the transcoding idea is correct.

Alok Jaiswal · ‎03-28-2012

Hi Anthony,

Have you tried the MTP option and set right hardware transcoder on the SIP trunk? Please also keep in mind to reset the SIP trunk to take the effect.

Also what i want to know is that when you setup a normal call between a Cisco 9971 or some endpoint registered on CUCM and dials to ex90 on VCS does call connects properly??/

Based on the logs what you have pasted i can point some more troubleshootig steps

First thing you will find in the logs is

2415.57 MC I: RemoteParticipant::reevalRefMode(p=11,ch=1) set ref [Audio

(1): G.711a 64k ] q= auto, t60=6000

which in my understanding is a indication for remote participant audio and i don't think i am wrong in thinking this as the call comes from PRI and it has to be G.711 codec.

Second thing

Under the 200OK going from the Ex90 i see the below logs.

2415.62 SipPacket m=audio 2386 RTP/AVP 100 102 103 104 105 106 9 8 0 101

2415.62 SipPacket b=TIAS:128000

2415.62 SipPacket a=rtpmap:100 MP4A-LATM/90000

2415.62 SipPacket a=fmtp:100

profile-level-id=25;object=23;bitrate=128000

2415.62 SipPacket a=rtpmap:102 MP4A-LATM/90000

2415.62 SipPacket a=fmtp:102 profile-level-id=24;object=23;bitrate=64000

2415.62 SipPacket a=rtpmap:103 MP4A-LATM/90000

2415.62 SipPacket a=fmtp:103 profile-level-id=24;object=23;bitrate=56000

2415.62 SipPacket a=rtpmap:104 MP4A-LATM/90000

2415.62 SipPacket a=fmtp:104 profile-level-id=24;object=23;bitrate=48000

2415.62 SipPacket a=rtpmap:105 G7221/16000

2415.62 SipPacket a=fmtp:105 bitrate=32000

2415.62 SipPacket a=rtpmap:106 G7221/16000

2415.62 SipPacket a=fmtp:106 bitrate=24000

2415.63 SipPacket a=rtpmap:9 G722/8000

2415.63 SipPacket a=rtpmap:8 PCMA/8000

2415.63 SipPacket a=rtpmap:0 PCMU/8000

2415.63 SipPacket a=rtpmap:101 telephone-event/8000

2415.63 SipPacket a=fmtp:101 0-15

2415.63 SipPacket a=sendrecv

its sending the G722 and MP4A codec which is not going to match with G.711 codec of PRI and you need a transcoder to transcode the G.722 to G.711 which can only be done on CUCM side by using a Hardware transcoders.

Third thing

I see that endpoint is advertising the BFCP? And if you see that ACK coming from the CUCM you will see there is no SDP present in it and i am assuming that you didn't strip any thing from it.

Also from the logs i can see you have CUCM 8.5. And i don't think CUCM 8.5 has support for BFCP.

So this could also be a reason...

2415.54 SipMedia W: SIPMEDIA_connectBFCP The BFCP medialine is not

negotiated

Thanks

Alok