Presence SIP in EX90

Similar Questions

• E20 hang with Sip Config - TE4.1.1 - Strange

  We see that some of our E20 - most of them on the internet is suspended as soon as endpoint is started.

  If the end point is without a network cable, then the E20 is stable.

  If the SIP configuration is deleted, the E20 is stable.

  As soon as we put in the SIP config and records the E20 on SIP, the unit freezes.

  We see root that enforcement Tandberg failure - root cli tsh says 'unable to connect request.

  Is this a known issue with TE4.1.1.

  If we disable SIP and only use H323 - device is stable.

  VCS - X7.2.1

  Hello

  the log files you sent from the E20 show a problem with the Marvell chip that is inside the E20 ethernet controller.

  Feb 26 07:59:02 (none) principal: FPGA programmed OK

  Feb 26 07:59:02 (none) main: marvell.c: ioctl (9,-2146669310,...) ==-1, errno == 14, wrong address

  Feb 26 07:59:02 (none) main: platform/marvell/marvell.c:132: marvell_ioctl: Assertion ' 0 & 'Marvell ioctl failed' ' failed.

  Feb 26 07:59:02 (none) main: signal received SIGABRT (6) wire 0x4082a4c0, 1704 TID

  Feb 26 07:59:02 (none) principal: records:

  Feb 26 07:59:02 (none) principal: R0: 00000000 R1: 000006a 8 R2: R3 00000006: 000006a 8

  Feb 26 07:59:02 (none) principal: R4: R5 00000006: 40826bdc R6: 40826000 R7: 0000010 c

  Feb 26 07:59:02 (none) principal: R8: 00000bdc R9: 00000000 R10: FP 4082a4c0: be976974

  Feb 26 07:59:02 (none) principal: PC: 407206f8 IP: be976978 SP: be97695c LR: 407206 c 4

  Feb 26 07:59:02 (none) principal: ERR: 00000000 CPSR: 20000010 FAULT: TRAP 00000000: 00000000

  Feb 26 07:59:02 (none) main: OLDMSK: 00000000

  Failure to our tracking system is down, so couldn't find a DDT corresponding to this, but to me, it sounds like the hardware.

  Will check with engineering.

  The recent change you speak may cause some E20s crashing. There is an open on this default:

  CSCue59199"target ="_blank"> CSCue59199 - Boot: error SIPAUTH: cannot delete the signature (gState = 1)"

  When we are challenged on a presence SIP subscribe message, the E20 crashes. When this happens, you will see a message like "SUBSCRIBE SUBSCRIBE got proxy-challenged in 407 authorization."

  Now the files of historical newspapers that sent you to the specific E20 does not match this DDT. You have other E20s crashing when you activate the SIP? If so, can you send files of historical log of such a device, which now no longer crashes with disabled SIP?

  Here's the sequence before the crash.

  Could you gather some debug SIP and try to activate the SIP on 1 E20 please?

  The tsh c:

  The ctx sippacket debug log 9

  When the aircraft crashes, disable SIP again and transfer of log files to check.

  3 Mar 11:43:43 (no) principal: admin user (1001) executed successfully the command ' / presence / to subscribe/Start URI: 91000001' sweet-infusion - 7.cisco.com.

  "3 Mar 11:43:43 (no) principal: 8241.76 SipStack I: SipEv: Active subscribe to"

  SIP: [email protected] / * /.

  ' type 'presence', unsolicited = 0

  3 Mar 11:43:43 (no) principal: 8241.76 I: SipUa added GRUU OK SipStack

  3 Mar 11:43:43 (no) principal: 8241.77 SipPacket PacketDump: Proto: SIP, name: SUBSCRIBE

  SIP: [email protected] / * /.

  SIP/2.0, Direction: Outgoing or remoteAddress is set: 10.106.93.69:5061, GroupEntity:

  [email protected] / * /.

  Time: 8241765 content:!

  3 Mar 11:43:43 (no) principal: 8241.77 SipPacket SUBSCRIBE

  SIP: [email protected] / * /.

  SIP/2.0

  3 Mar 11:43:43 (no) principal: 8241.77 SipPacket Via: SIP/2.0/TLS 171.69.87.88:5061; branch = z9hG4bKb13dc154a422586a9cb016c9d7ac0a3f.1; rport

  3 Mar 11:43:43 (no) principal: 8241.77 SipPacket Call-ID:

  [email protected] / * /.

  3 Mar 11:43:43 (no) principal: 8241.77 SipPacket CSeq: SUBSCRIBE 101

  3 Mar 11:43:43 (no) principal: 8241.78 SipPacket Contact:

  3 Mar 11:43:43 (no) principal: 8241.78 SipPacket to:

  ; tag = 92c18ad11538b1bd

  3 Mar 11:43:43 (no) principal: 8241.78 SipPacket to:

  3 Mar 11:43:43 (no) principal: 8241,78 SipPacket Max-Forwards: 70

  3 Mar 11:43:43 (no) principal: road of SipPacket of 8241.78:

  3 Mar 11:43:43 (no) principal: 8241.78 SipPacket User-Agent: TANDBERG/257 (TE4.1.1.271887Beta1)

  3 Mar 11:43:43 (no) principal: 8241.78 SipPacket Expires: 3600

  3 Mar 11:43:43 (no) principal: 8241.78 SipPacket event: presence

  3 Mar 11:43:43 (no) principal: 8241.78 SipPacket Accept: application/pidf + xml

  3 Mar 11:43:43 (no) principal: 8241.78 SipPacket Content-Length: 0

  3 Mar 11:43:43 (no) principal: SipPacket 8241.78

  3 Mar 11:43:43 (no) principal: SipPacket 8241.78 >!

  3 Mar 11:43:43 (no) principal: 8242.02 SipPacket PacketDump: Proto: SIP, name: SIP/2.0 407 Proxy Authentication Required, Direction: inbound, RemoteAddress: 10.106.93.69:5061, GroupEntity:

  [email protected] / * /.

  Time: 8242018 content:!

  3 Mar 11:43:43 (no) principal: 8242.02 SipPacket SIP/2.0 407 Proxy Authentication Required

  3 Mar 11:43:43 (no) principal: 8242.02 SipPacket Via: SIP/2.0/TLS 171.69.87.88:5061; branch = z9hG4bKb13dc154a422586a9cb016c9d7ac0a3f.1; received = 171.69.87.88; rport = 36924

  3 Mar 11:43:43 (no) principal: 8242.02 SipPacket Call-ID:

  [email protected] / * /.

  3 Mar 11:43:43 (no) principal: 8242.03 SipPacket CSeq: SUBSCRIBE 101

  3 Mar 11:43:43 (no) principal: 8242.03 SipPacket to:

  ; tag = 92c18ad11538b1bd

  3 Mar 11:43:43 (no) principal: 8242.03 SipPacket to:

  ; tag = aaa367e278a7ece0

  3 Mar 11:43:43 (no) principal: 8242.03 SipPacket server: TANDBERG/4120 (X7.2.1)

  3 Mar 11:43:43 (no) principal: 8242,03 SipPacket Proxy-Authenticate: Digest realm = "akgvcsc1.ciscolab.com", nonce = "bb98866b8387ba58270573abceefb75e19dcde22e51b036493413ef0b5cd", opaque = "AQAAAK5Mba8WRqO56xvFJpIWzjCx51zZ", stale = FALSE, algorithm = MD5, qop = "auth"

  3 Mar 11:43:43 (no) principal: 8242.04 SipPacket Content-Length: 0

  3 Mar 11:43:43 (no) principal: SipPacket 8242.04

  3 Mar 11:43:43 (no) principal: SipPacket 8242.04 >!

  3 Mar 11:43:43 (no) principal: permission to SUBSCRIBE got proxy-challenged in 407 SUBSCRIBE

  3 Mar 11:43:43 (no) main: signal received SIGSEGV (11) wire 0x4084d450, TID 1809

  3 Mar 11:43:43 (no) principal: illegal memory access to: 0x8

  3 Mar 11:43:43 (no) principal: records:

• Accommodation of the presence on the next information control VCS

  Hello!
  Could you please help me?

  What should I do to host the presence information of my domain name SIP not on local control of VCS, where points registered endpoint, but VCS control located in the nearby area?

  Disable the server of presence on your own VCS control while allowing the presence on VCS neighbored control server.

  Also make sure that your local SIP domain exists also on VCS neighbored control and that the rules of research between these VCS allow presence SIP traffic pass freely between the two (AnyAlias for example).

  -Andreas

• Max calls - Material Limitation on EX90?

  I sent 2 x EX90s CUCM and find the max calls on the DN for the EX90 is 4.  Without implementing this to MCU, that someone knows or can confirm it is indeed hardware limitation?

  SIP CiscoTelepresence EX90 - DN has 4 maximum of calls.

  EX90 running TC6.2

  UCM - 9.1

  SCCP phones - DN have 200 calls maximum.

  Kind regards

  John

  Hi John,.

  Using her own from the embedded resource multipoint EX90 is able to make the 4 at the same time calls, taped to CUCM or not. However, to enable this feature, you must have the 'Multisite Support' (LIC-EX90-MS) license.

  And Yes, it is a hardware limitation, you can have up to 4 participants 720p30 connected to your EX90.

  Take a look at the EX series datasheet:

  http://www.Cisco.com/en/us/prod/collateral/ps7060/ps11303/ps11308/ps11327/data_sheet_c78-627494.html
  CTRL + F and search for "Multisite"

  I hope this helps.

  Concerning

  Paulo Souza

  My answer was helpful? Please note the useful answers and do not forget to mark questions resolved as "responded."

• How can I set the resolution to 1080 p using SIP H.323 or SIP protocols H.323 by VCS on EX90, C60 both?

  Hi, I'm an engineer junior network in Korea.

  and, for the first time, I apologize for my bad writing to read in uncomportable. Sorry, as eng is not my mother tongue.

  I tested on SIP to SIP and H.323 to H.323 call. These cases are all correctly, in 1080P(1920*1080),

  BUT

  SIP to H.323 or H.323 SIP dosen't work through VCS. The types of devices are EX90, C60.

  Status of devices: HD-resolution activated license, 6000K, the default bandwidth

  and there is no external problems, because it is the TURN of test on the same LAN.

  I tried several times, but these devices connect sometimes 720P(1280*720) and sometimes each other 1080 p resolution.

  Is anyone knows how to put the devices in trouble their 1080 p not 720 p resolution?

  I'll look forward to your good kindness.

  Please read my bad writing.

  Jinsung salvation.  It would be important to know what SW version running on the CV.  Before X7.1, VCS would set lines for SIP step of BW calling in 1920, and this is fixed in this version.  I would like to ask, if you can, upgrade your VCS to X7.1 and try the call again.  It is perhaps what is you covering for the moment and achieve 1080 p on Interworking call.

  CSCtx32717

  Symptoms: A call interworked will find its video bandwidth, capped at 2 Mbps in one direction.

  Conditions: Video sent by the side of the SIP in the H323 will be capped at 2 Mbps. This may cause a different codec being chosen, or a lower bandwidth on the favorite codec.

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/VCs/release_note/Cisco_VCS_Release_Note_X7-1.PDF

  You can try:

  Workaround would be to set optimal definition to high on both
  
  endpoints or avoid interworking to achieve 1080p in the meantime.
  
  Let us know the outcome.
  Thanks in advance.
  VR
  
  P2

• Add EX90 cm (sip)

  We run AAU 8.6.1.2006

  EX90 TC7.3.0

  Impossible to get the EX90 register with the CM?

  Error endpoint is supply - HTTP-404

  Any help is appreciated...

  Checked connectivity Provisioning is Auto

  HttpMethod is GET

  SIP status is inactive

  Hi Chet,

  A few points:

  1. check if the directory number has been configured on CUCM if you add manually.

  2 CUCM 8.6.2 or higher is recommended according to the EX series datasheet.

  Manish

• EX90 two autonomous with the public IP address can make video calls among them self on the Internet or not?

  Dear expert;

  I am very new to VCS and TP Cisco.

  We implement now presence Cisco TV with VCS - C, VCS-E TMS, TCS, MCUS and endpoints with Jabber in a single edit.

  and in another configuration CUCM 10.5, UCCX 10.5 IM & P, Jabber with some 10 officers.

  Now the question is in our building on the 2nd floor we have an EX90 and on the 5th floor an EX90 and on local network, we can make video calls using the IP address.

  In the same way is it possible to make a video call between 2 devices EX90 (both have public IP) present in a location different in the same city on the Internet without the participation of VCS - C and VCS-E.

  It's the client request :)

  Concerning

  Paiva

  Yes, but leaving these systems outside in nature with public IP addresses, leaving you are vulnerable to a number of questions. See for example http://www.videonationsltd.co.uk/2014/11/h-323-cisco-spam-calls/

  https://supportforums.Cisco.com/discussion/12336591/sourceh323idcisco-incomingcalls

  https://supportforums.Cisco.com/discussion/12340591/nuisance-h323-calls-SX20

  The offers above with H.323 calls, in addition to this, you will encounter similar problems using SIP where the systems will be analyzed by tools such as SIPVicious

  /Jens

  Please note the answers and mark questions as "answered" as appropriate

• SIP message with 488 here is not Acceptable

  Hello, I'm having this problem. I have read the documentation and I look in this forum for similar problems.
  Seems is a Codec problem. Service provider asks a g711alaw. However, I already change the codec, transcoder configured which is register with handler calls and everything seems fine. But outgoing calls always give this error. Can someone help me?

  Here's the part of my configuration:

  Send-call voice alert

  convert-discpi-to-prog voice calls

  voice, send rtp-received

  voip phone service

  list of approved IP addresses

  IPv4 10.20.1.0 255.255.255.0

  h323 connections allow h323

  allow connections h323 to SIP

  allow connections sip h323

  allow sip to sip connections

  service additional h450.12

  Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none

  H323

  No keepalive timeout h225

  SIP

  midcall-signalling passthru

  SCCP local GigabitEthernet0/1

  SCCP ccm 10.20.1.5 identifier 2 priority 2 version 7.0

  SCCP ccm 10.20.1.6 identifier 1 priority 1 version 7.0

  SCCP

  !

  SCCP ccm Group 1

  associate the ccm 1 priority 1

  the associated profile 1 registry COMPcode

  the associated profile 2 registry COMP01-PSG

  !

  transcode dspfarm profile 1

  Codec g711ulaw

  Codec g711alaw

  Codec g729ar8

  Codec g729abr8

  Codec g729r8

  Codec g729br8

  maximum sessions 14

  associate the PCRS application

  !

  dspfarm profile 2 PSG

  Codec g711ulaw

  maximum sessions 120 software

  associate the PCRS application

  !

  Dial-peer voice voip 50

  Description # calls of CUCM to VG #.

  incoming called number 9.T

  DTMF-relay h245 alphanumeric

  Codec g711alaw

  No vad

  !

  Dial-peer voice 11 voip

  Description * outgoing SIP Trunk call *.

  outgoing SIP CALLS OUT translation-profile

  preference 1

  destination-model 9 t

  Setup progress_ind allow 3

  progress_ind enable progress 8

  progress_ind connect enable 8

  redirect ip2ip

  session protocol sipv2

  session target ipv4:88.XXX. XX.XXX

  DTMF-relay rtp - nte cisco-rtp

  Codec g711alaw

  No vad

  The Sip Trunk on the call manager uses a MRGL both of PSG and XCODE.

  SCCP Admin State: to the TOP
  Gateway local Interface: GigabitEthernet0/1
  IPv4 address: 10.20.1.11
  Port number: 2000
  IP precedence: 5
  List of Codec hidden user: no
  Call Manager: 10.20.1.5, Port number: 2000
  Priority: 2, Version: 7.0, identifier: 2
  Call Manager: 10.20.1.6, Port number: 2000
  Priority: 1, Version: 7.0, identifier: 1

  Transcoding of Oper status: ACTIVE - reason Code: NO

  Here are the two him debugs:

  * 3 Dec 12:08:32.859: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  GUEST sip:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/TCP 10.20.1.6:5060; branch = z9hG4bK148c5e3b74ba77
  From: <> [email protected]/ * */>;tag=1130165~6e3f6e1b-b59e-4604-be6a-d6295dc18cf6-45522220
  To: <> [email protected]/ * / >
  Date: Monday, December 3, 2012 12:02:40 GMT
  Call ID: [email protected] / * /
  Supported: timer, resource-priority, replaces
  Min - SE: 1800
  User-Agent: Cisco - CUCM8.6
  Allow: PROMPT, OPTIONS, INFO, BYE, ACK, CANCEL, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY
  CSeq: INVITE 101
  Expires: 180
  Allow-events: presence, kpml
  Support: X-cisco-srtp-relief
  Support: geolocation
  Call-Info: ; method = "NOTIFY; Telephone-event = event; duration = 500 "
  Cisco-Guid: 1422849408-0000065536-0000391810-0100733962
  Session time-out: 1800
  P has asserted-Identity: <> [email protected]/ * / >
  Remote-Party-ID: <> [email protected]/ * / >; left = call; screen = yes; intimacy = off
  Contact: <> [email protected]/ * /: 5060; transport = tcp >
  Max-Forwards: 70
  Content-Length: 0

  * 3 Dec 12:08:32.863: //-1/54CEF5800005/CCAPI/cc_api_display_ie_subfields:
  cc_api_call_setup_ind_common:
  Cisco-username = 02036666666
  -ccCallInfo IE subfields-
  Cisco-ani = 02036666666
  Cisco-anitype = 0
  Cisco-aniplan = 0
  Cisco-anipi = 0
  Cisco-anisi = 1
  dest = 907718005555
  Cisco-desttype = 0
  Cisco-destplan = 0
  Cisco-ISDS = FFFFFFFF
  Cisco-rdn =
  Cisco-rdntype = 0
  Cisco-rdnplan = 0
  Cisco-rdnpi =-1
  Cisco-rdnsi =-1
  Cisco-redirectreason = - 1 fwd_final_type = 0
  final_redirectNumber =
  hunt_group_timeout = 0

  * 3 Dec 12:08:32.863: //-1/54CEF5800005/CCAPI/cc_api_call_setup_ind_common:
  Interface = 0 x 31196458, call Info)
  Number = 02036666666, (Calling Name =) (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed).
  Called number = 907718005555 (TON = unknown, NPI = unknown).
  The appeal translated = FALSE, Subscriber Type Str = unknown, FinalDestinationFlag = TRUE,
  Incoming dial-peer = 50, progress Indication = NULL (0), Calling THE Present = TRUE,
  Road Trkgrp Label source, label Trkgrp road target = CLID Transparent = FALSE), call Id = 21355
  * 3 Dec 12:08:32.863: //-1/54CEF5800005/CCAPI/ccCheckClipClir:
  In: Component number = 02036666666 (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed)
  * 3 Dec 12:08:32.863: //-1/54CEF5800005/CCAPI/ccCheckClipClir:
  Departures: Component number = 02036666666 (TONE = unknown, NPI = unknown = User, spent, screening presentation allowed =)
  * 3 Dec 12:08:32.863: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

  * 3 Dec 12:08:32.863: cc_get_feature_vsa success of malloc
  * 3 Dec 12:08:32.863: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

  * 12:08:32.863 3 dec: cc_get_feature_vsa number is 1
  * 3 Dec 12:08:32.863: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

  * 12:08:32.863 3 dec: FEATURE_VSA attributes are: feature_name:0, feature_time:832856160, feature_id:21306
  * 12:08:32.863 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_api_call_setup_ind_common:
  Set up the event sent;
  Call Info (number = 02036666666 (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed),)
  Called number = 907718005555 (TON = unknown, NPI = unknown))
  * 12:08:32.863 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_process_call_setup_ind:
  Event = 0x2AFCEB88
  * 3 Dec 12:08:32.863: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
  Try again with the demoted called number 907718005555
  * 12:08:32.863 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallSetContext:
  Context = 0x32D52B8C
  * 12:08:32.863 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_process_call_setup_ind:
  > Handed CCAPI cid 21355 with tag 50 app '_ManagedAppProcess_Default '.
  * 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallProceeding:
  Progress Indication = NULL (0)
  * 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallSetupRequest:
  Destination =, Calling IE date = TRUE, Mode = 0.
  Leaving Dial-peer = 11, Params = 0x2B62716C, Indication of progress = FROM SIDE IS NO ISDN (3)
  * 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCheckClipClir:
  In: Component number = 02036666666 (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed)
  * 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCheckClipClir:
  Departures: Component number = 02036666666 (TONE = unknown, NPI = unknown = User, spent, screening presentation allowed =)
  * 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallSetupRequest:
  The destination model = 9 t., called number = 07718009863, band numbers = FALSE
  * 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallSetupRequest:
  Number = 02036666666 (TON = unknown, NPI = unknown, screening = User, spent, presentation = allowed).
  Called number = 07718009863 (TON = unknown, NPI = unknown).
  Redirect = number, display of information is
  Account number 02036666666, Destination = final = TRUE flag,.
  GUID = 54CEF580-0001-0000-0005-FA820601140A, outbound Dial-peer = 11
  * 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_api_display_ie_subfields:
  ccCallSetupRequest:
  Cisco-username = 02036666666
  -ccCallInfo IE subfields-
  Cisco-ani = 02036666666
  Cisco-anitype = 0
  Cisco-aniplan = 0
  Cisco-anipi = 0
  Cisco-anisi = 1
  dest = 07718009863
  Cisco-desttype = 0
  Cisco-destplan = 0
  Cisco-ISDS = FFFFFFFF
  Cisco-rdn =
  Cisco-rdntype = 0
  Cisco-rdnplan = 0
  Cisco-rdnpi =-1
  Cisco-rdnsi =-1
  Cisco-redirectreason = - 1 fwd_final_type = 0
  final_redirectNumber =
  hunt_group_timeout = 0

  * 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccIFCallSetupRequestPrivate:
  Interface = 0 x 31196458, Type of Interface = 3, = Destination, the Mode = 0x0,
  Call Params (number = 02036666666, (Calling Name =) (= unknown, NPI = unknown, screening = User, TON spent, presentation = allowed),)
  Called number = 07718009863 (TON = unknown, NPI = unknown), the appeal translated = FALSE,
  Subscriber Type Str = unknown, FinalDestinationFlag = TRUE, outgoing Dial-peer = 11, Call On County = FALSE,
  Trkgrp road Label source, label of road target Trkgrp =, tg_label_flag = 0, Application Call Id =)
  * 3 Dec 12:08:32.867: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

  * 3 Dec 12:08:32.867: cc_get_feature_vsa success of malloc
  * 3 Dec 12:08:32.867: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

  * 12:08:32.867 3 dec: cc_get_feature_vsa number is 2
  * 3 Dec 12:08:32.867: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:

  * 12:08:32.867 3 dec: FEATURE_VSA attributes are: feature_name:0, feature_time:832857952, feature_id:21307
  * 12:08:32.867 3 dec: / / 21356, 54CEF5800005, CCAPI, ccIFCallSetupRequestPrivate:
  Application for facility call SPI's success; Interface type = 3, FlowMode = 1
  * 12:08:32.867 3 dec: / / 21356, 54CEF5800005, CCAPI, ccCallSetContext:
  Context = 0x2B62711C
  * 12:08:32.867 3 dec: / / 21355, 54CEF5800005, CCAPI, ccSaveDialpeerTag:
  Outbound Dial-peer = 11
  * 12:08:32.867 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_proceeding:
  Interface = 0 x 31196458, Indication = NULL (0) progress
  * 12:08:32.871 3 dec: / / 21356/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
  Envoy:
  GUEST sip:[email protected] / * /.XXX:5060 SIP/2.0
  Via: SIP/2.0/UDP 194.168.146.148:5060; branch = z9hG4bK4F1C77
  Remote-Party-ID: <> [email protected]/ * / >; left = call; screen = yes; intimacy = off
  From: <> [email protected]/ * / >; tag = E448A58-A24
  To: <> [email protected] / * /.XXX >
  Date: Monday, December 3, 2012 12:08:32 GMT
  Call ID: [email protected] / * /
  Supported: timer, resource-priority, replaces, sdp-anat
  Min - SE: 1800
  Cisco-Guid: 1422849408-0000065536-0000391810-0100733962
  User-Agent: Cisco-SIPGateway/IOS - 12.x
  Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
  CSeq: INVITE 101
  Time stamp: 1354536512
  Contact: <> [email protected]/ * /: 5060 >
  Expires: 180
  Allow-events: telephone-event
  Max-Forwards: 69
  Session time-out: 1800
  Content-Length: 0

  * 12:08:32.871 3 dec: / / 21355/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
  Envoy:
  SIP/2.0 100 trying
  Via: SIP/2.0/TCP 10.20.1.6:5060; branch = z9hG4bK148c5e3b74ba77
  From: <> [email protected]/ * */>;tag=1130165~6e3f6e1b-b59e-4604-be6a-d6295dc18cf6-45522220
  To: <> [email protected]/ * / >
  Date: Monday, December 3, 2012 12:08:32 GMT
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Allow-events: telephone-event
  Server: Cisco-SIPGateway/IOS - 12.x
  Content-Length: 0

  * 12:08:32.899 3 dec: / / 21356/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
  Received:
  SIP/2.0 100 trying
  Via: SIP/2.0/UDP 194.168.146.148:5060; received = 194.168.146.148; branch = z9hG4bK4F1C77
  From: <> [email protected]/ * / >; tag = E488A58-A24
  To: <> [email protected] / * /.XXX >
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Time stamp: 1354536512
  Content-Length: 0

  * 12:08:32.995 3 dec: / / 21356/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
  Received:
  SIP/2.0 488 here is not Acceptable
  Via: SIP/2.0/UDP 194.168.146.148:5060; received = 194.168.146.148; branch = z9hG4bK4F1C77
  To: <> [email protected] / * /.XXX >; tag = 3563525093-836070
  From: <> [email protected]/ * / >; tag = E488A58-A24
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
  Contact: <> [email protected] / * /.XXX:5060 >
  Reason: Q.850; cause = 65
  Content-Length: 0

  * 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_disconnected:
  Value = 127, Interface = 0 x 31196458, Call Id = 21356
  * 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_disconnected:
  Call the entry (Responsed = TRUE, Cause value = 127, Retry Count = 0)
  * 12:08:32.995 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallReleaseResources:
  free xcoding reserved resource.
  * 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, ccCallSetAAA_Accounting:
  = 0, Id = 21356 Call Accounting
  * 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, ccCallDisconnect:
  Value = 127, Tag = 0x0, entry calls (previous disconnection Cause = 0, remove the Cause = 127)
  * 12:08:32.995 3 dec: / / 21356, 54CEF5800005, CCAPI, ccCallDisconnect:
  Value = 127, entered calls (Responsed = TRUE, Cause value = 127)
  * 12:08:32.999 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_disconnect_done:
  Available = 0, Interface = 0 x 31196458, Tag = 0 x 0, Call Id 21356 =.
  Call the entry (disconnect Cause = 127, class voice Cause Code = 0, retry count = 0)
  * 12:08:32.999 3 dec: / / 21356, 54CEF5800005, CCAPI, cc_api_call_disconnect_done:
  Call interrupted event sent
  * 3 Dec 12:08:32.999: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

  * 3 Dec 12:08:32.999: cc_free_feature_vsa release A 31, 46758
  * 3 Dec 12:08:32.999: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

  * 3 Dec 12:08:32.999: free vsacount is 1
  * 12:08:32.999 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallDisconnect:
  Value = 127, Tag = 0x0, entry calls (previous disconnection Cause = 0, remove the Cause = 0)
  * 12:08:32.999 3 dec: / / 21355, 54CEF5800005, CCAPI, ccCallDisconnect:
  Value = 127, entered calls (Responsed = TRUE, Cause value = 127)
  * 12:08:32.999 3 dec: / / 21355/54CEF5800005/SIP/Msg/ccsipDisplayMsg:
  Envoy:
  SIP/2.0 500 Internal Server Error
  Via: SIP/2.0/TCP 10.20.1.6:5060; branch = z9hG4bK148c5e3b74ba77
  From: <> [email protected]/ * */>;tag=1130165~6e3f6e1b-b59e-4604-be6a-d6295dc18cf6-45522220
  To: <> [email protected]/ * / >; tag = E488ADC-CEC
  Date: Monday, December 3, 2012 12:08:32 GMT
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Allow-events: telephone-event
  Server: Cisco-SIPGateway/IOS - 12.x
  Reason: Q.850; cause = 127
  Content-Length: 0

  * 3 Dec 12:08:33.035: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  SIP ACK:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/TCP 10.20.1.6:5060; branch = z9hG4bK148c5e3b74ba77
  From: <> [email protected]/ * */>;tag=1130165~6e3f6e1b-b59e-4604-be6a-d6295dc18cf6-45522220
  To: <> [email protected]/ * / >; tag = E488ADC-CEC
  Date: Monday, December 3, 2012 12:02:40 GMT
  Call ID: [email protected] / * /
  Max-Forwards: 70
  CSeq: 101 ACK
  Allow-events: presence, kpml
  Content-Length: 0

  * 12:08:33.039 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_api_call_disconnect_done:
  Available = 0, Interface = 0 x 31196458, Tag = 0 x 0, Call Id = 21355.
  Call the entry (disconnect Cause = 127, class voice Cause Code = 0, retry count = 0)
  * 12:08:33.039 3 dec: / / 21355, 54CEF5800005, CCAPI, cc_api_call_disconnect_done:
  Call interrupted event sent
  * 3 Dec 12:08:33.039: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

  * 3 Dec 12:08:33.039: cc_free_feature_vsa release A 31, 46058
  * 3 Dec 12:08:33.039: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:

  * 3 Dec 12:08:33.039: free vsacount is 0
  * 3 Dec 12:08:33.039: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Envoy:
  SIP ACK:[email protected] / * /.XXX:5060 SIP/2.0
  Via: SIP/2.0/UDP 194.168.146.148:5060; branch = z9hG4bK4F1C77
  From: <> [email protected]/ * / >; tag = E488A58-A24
  To: <> [email protected] / * /.XXX >; tag = 3563525093-836070
  Date: Monday, December 3, 2012 12:08:32 GMT
  Call ID: [email protected] / * /
  Max-Forwards: 70
  CSeq: 101 ACK
  Allow-events: telephone-event
  Content-Length: 0

  Thank you very much

  Chris

  Hi Chris,

  It seems that your CUBE config is correct. The CUCM sends a PROMPT to provide at the beginning now. This means that you have enabled the PSG on the SIP trunk. The codec sent in the INVITATION is G711ulaw. In order to change this, go to the SIP trunk and search "PSG preferred from Codec" and replace it by default of G711ulaw, G711alaw. If all goes well, which does the trick for you.

  HTH.

  Kind regards

  Stefano.

• Configure SIP URI dialing

  Hello

  I tried to activate the SIP URI dial, unfortunately with no success yet. Here you can see what I've done, maybe one of you can tell me what is missing:

  • Set up the LDAP synchronization, assign msRTCSIP-PrimaryUserAddress URI directory (because of the interoperability of Lync)
  • DNs associated with devices of CSF and BOT
  • Associate the URIs directory with directory numbers
  • The "phones" partition of directory URI alias value
  • Modify a SIP profile (named Jabber) later
    • Define interpreting string dial with 0-9, *, # and +.
    • Use the FULL domain name in the SIP application
    • Allow the presentation of bfcp
  • Assign the profile Jabber SIP devices CSF and BOT
  • Set the DNs route partition as an alias of the directory URI (phones)
  • Jabber - config.xml and change download
    • msrtcsip-primaryuseraddress in section of the book
    • true in section strategies
  • Restart the tftp server and re - open a session with Jabber

  We have 9.1.2.11900 - 12 Call Manager Server Instant Messaging and presence 9.1.1.10000 - 8 and 9.6/9.7 Jabber clients.

  Is there anything I missed or misconfigured?

  Thank you

  LACI

  Hi Labelle,

  If you put DirectoryURI in the uri section directory publication is no longer available in the client.

  Of the documentention, I got that it belongs to the policies. But if I put it here my click to call URI is always the address e-mail and not msRTCSIP-PrimaryUserAddress. So I'll do some research here.

  One thing I found is the resolution of contact section. If I signed to jabber my name did not appear. Instead, it was my msRTCSIP-PrimaryUserAddress.

  To resolve this problem, you will need to add this to jabber - config.Xml.

  true
    msRTCSIP-PrimaryUserAddress
    SIP:

  Kind regards

  Paul

• Cisco EX90: Unable to see the presentation of the end

  Hi all

  We have currently enrolled in CUCM EX90.

  Question:

  Scenario of

  1 when the EX90 will launch to share the presentation, from the other side, they were able to see the presentation that we shared.

  but in the far endpoint if they will start to share the presentation we can see their presentation and this is the only time wherever they are able to see our shared presentation.

  Conclusion: before we able to share the presentation at the other end, firstly they must commit to share their presentation we before they can see our presentation.

  Already checked the NaturalPresenter but according to audit it is already built in the EX90.

  My question is why the presentation works after end first start sharing their screen for us. But if we will undertake the first, they can't see our split screen.

  Receive your answer. Thank you!

  It will depend on the version of the software you; "re running on the EX90.

  Before TC7.0 EX90 Admin Guide says that, for a call SIP and the 5170-5177 5070-5077 UDP ports are used to connect BFCP (channel of submission).  From TC7.0 onwards, the BFCP port and the active control prefer to use dynamically ports the RTP port range.

  If a firewall blocks these ports, channel of submission will not be established and the presentation can be sent in the channel video main, which is probably why it seems to work when you call a bridge.

  Note: It is strongly recommended that you upgrade your version of software on your EX90 to TC7.3.6 to address many security vulnerabilities in earlier software releases.

  Wayne
  --
  Remember the frequency responses and mark your question as answered as appropriate.

• Unable to connect to presence server Jabber clients

  Salvation;

  I am a newbie in Collaboration and to learn basics who set up a small laboratory including 11 CUCM, presence 11 and two phones 7821 Cisco. I have configured CUCM and presence and set up a SIP trunk between them according to the documents. I tried to configure the client Jabber on Windows, but in vain. first of all, I created some local users on CUCM and have some LDAP users that have been synchronized with a test AD in which both the presence and the CUCM are members of this domain. I set up a local users CUCM for Jabber Client and to configure as follows to configure the Jabber Clients according to the documents.

  My first question is the format of the user name that should be used with Jabber. client Jabber said we use "[email protected] / * /" format, but I saw the videos on the Internet using normal user name and password. regarding the fact that I have configured a local user for the Jabber CUCM, I entered in the form of "username" but Jabber said the format should be as "[email protected] / * /'." "

  the second problem is that even after writing the user name in this format, the Jabber said that 'not to communicate with the server'. I can ping the CUCM and presence server both by using their ip addresses and hostnames through my PC on which is installed the Jabber. also I wrote serves him the IP address on the Jabber client while trying to connect the Jabber instead of their host names. is there any requirement to be member of a single domain?

  I tried Jabber for iPad to connect to the presence, but in vain too. y at - it knows no question or tips about the present? Otherwise, I can explain the steps I went through them to set up the presence and the CUCM for Jabber clients.

  best regards;

  Because for the most part, allows you to know the details of connection SRVs and SRVs work based on the field.

  You have successfully configured your SRVs according to configuration guides?

  It is very likely that some missing config or a misconfiguration.

  I have videos on how to configure the IM & P as well as Jabber config integration

  https://www.YouTube.com/playlist?list=PLFuOESqSTxEvZChqWgAJanctohRMe99CR

• Where can I find documentation on how to get the VCS - C and VCS-E to Exchange presence information

  I turned presence on my VCS - C and VCS-E, but depending on whether I am VPNed or not, I don't see the State of the presence of my side of the firewall.

  I checked for x7.1 Administrator's guide, I looked in the guide of authentication devices, and I'm at a bit of a loss.

  Is what I'm trying to do as possible, and if so, what do I have to do to make this work?

  I have a pretty simple setup - that works very well, could give you some ideas;

  Two VCS - Independent C (soon to be clustered) and a VCS-E;

  all internal clients of the JabberVideo register with a VCS - C, and when external registers with the VCS-E (all of my users of Singapore recorded with the VCS-E).

  Authentication is directly with AD and is made on the a WHAT VCS - C regardless of the user is registered with the VCS - C or the VCS-E.

  PUA and presence disabled on VCS-E and activated on the VCS - C, which is used as SIP registrar.

  Authentication on VCS - e:

  Default subfield "do not check".

  Default zone "do not check".

  Traversal server box "do not check".

  Authentication on VCS - C:

  Default subfield "audit powers.

  MOVI area 'audit powers.

  Point final SIP registration box "treat as authenticated.

  Course customer area 'audit powers.

  Authentication works well, the users cannot sign up for the VCS-E with the wrong password or user name, and presence works well too - both inside and outside.

  /Jens

• Problem with a SIP Trunk between VCS and CUCM

  Hello world

  I created a SIP Trunk between control VCS (X7.1) and a CUCM 8.6.

  I followed this Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_6-1_7_8_and_X7 - 0.pdf deployment guide.

  The trunk is in place, but I have the following problem: I can ring since an EX90 recorded on the VCS for a registered on the CUCM 9951 but when I take the call, I have a busy signals (audio and video). When I on the EX90 9951 ring, I have the following message on VC: can not reach the contact.

  I have a 503 service unavailable status on the call history and "call rejected" on the event log.

  My transformation is OK and the call is sent to my neighbor CUCM zone

  You have an idea to solve this problem?

  Thanks for your help

  Best regards

  Bruno Z.

  Hello

  Yes, it's true! It seems at first glance a problem with codec negotiation when I see the reason 47 code.

  in any case set the region to G.711 trunk phone in case if you have different device pool and the trunk region and ip-phones.

  make G.711 end to end and test again.

  Thank you

  Alok

• EX90 directory provisioning model

  Hello everyone,

  I just check the possibilities to provide endpoint as EX90.

  We have the VCS Control and - Expressway and a TMS (via TMSPE). VCS-Highway I'm proxy SIP registrations. H.323 registrations are on VCS Expressway. I created two models, one with an internal configuration and the other with an external configuration and different users.

  The only thing I changed on the external model is the address of the server SIP and H.323 Gatekeeper.

  When the Government of EX90 internal LAN, the directory is now available.

  But when I register of HomeOffice thanks to the internet, no directory is available. Registration with SIP and H.323 is ok, but no directory is displayed.

  On MSDS, I already checked if the directory is available for both models. Because with Jabber/Movi, the directory is available in recordings transmitted by proxy, I had hope, it is also possible with endpoints. Am I wrong?

  Best regards, Daniel

  The reason is that it does not get synced with before TMS. Forcing full synchronization may also have then solved the problem.

• Assistant provisioning EX90

  ON the EX90s, when I select the VCS Cisco as the option of Infrastructure, he invited me to

  Username

  Password

  Field

  and

  External Manager.

  What username and password it tries to authenticate? Where is this configured? What IS happening in TMS? I tried to create some configuration options in TMS similar to those I would create for Jabber, but the EX90 never pull a config.

  I can simply add the VCS-highway as proxy SIP EX90 configuration and it will record and place and receive calls very well, but I try to get the device to authenticate in fact.

  I looked at the device provisoning guide and the guide of authenticating devices, and this topic does not seem to be addressed. Does anyone have pointers or suggestions, they can share?

  Thank you, in advance,

  RF

  The details of the account under system > Provisioning > users

  We have the following parameters at the moment, but it's specific deployment depending on how you want to work the things - is not not a recommendation of the best way of doing things, because it probably but it works for our small office:

  Conference AutoAnswer Mode	Off
  Conference DefaultCall Protocol	SIP
  H323 LoginName authentication profile
  H323 Mode of authentication profile	On
  H323 password authentication profile
  H323 Mode of CallSetup for profile	Access controller
  H.323 Gatekeeper profile address
  H323 H323Alias profile ID	{username} .ex90 @.
  NetworkServices multi address	{username} .meetme @ (we have a conductor who takes those calls)
  Address NetworkServices NTP
  Type of phone book server	VCS
  The phone book server URL	Directory @.
  Provisioning ExternalManager address
  Field of supply ExternalManager
  Method of preparation	VCS
  Public SIP server address
  DefaultTransport SIP profile	TLS
  Outgoing SIP profile	Off
  The SIP server address
  Time zone	GMT (Greenwich mean time: Dublin, Edinburgh, Lisbon, London)