Rogue SIP calls on C40

Similar Questions

• Call SIP with Tandberg C40

  I have a Cisco Tandberg C40 codec and have been making H.323 calls for an organization with no problems.  The organization is now making changes and demanding that I connect to them with their new SIP address.  When I try to connect, it is said immediately that the call could not be connected.  What are the requirements for placing a SIP call with the Tandberg C40?

  Assuming your C40 has the version of the TC6.x software;

  go to network found under Control Panel Services and enable SIP.

  Don't forget of check that SIP is selected as the Protocol to call when calling this company that the default calling Protocol can be set to H.323, SIP and H.320, so if only make you SIP calls, and then upgrade the SIP, but if you do mostly H.323 calls, then put to H.323, and select SIP among the options of appeal when the dial - or even more easily, add this address to the local directory with specified as a SIP call Protocol.

  Just turning on SIP will allow you to make outgoing SIP calls, however, if this company wants to call you, then they can call you only using the public IP address of your system - and they could even not able to do according to their deployment, policies, etc.

  If you want people to be able to call you using a SIP URI as [email protected] / * /, then you need to register your system with a Registrar SIP CUCM, VCS - C, etc. and also have SRV SIP records in place.

  See the guide to the administrator for more information:

  http://www.Cisco.com/en/us/docs/Telepresence/endpoint/codec-c-series/TC4/administration_guide/profile-c60_codec-C60-c40_administrator_guide_tc40.PDF

  /Jens

  Please note the answers and score the questions as "answered" as appropriate.

• SIP calls failing

  Hello

  I've implemented VCS in both sites, Site A and site B and I did the neighbors between Site A and Site B VCS rules. Search and transformations are configured and implemented and applied.

  Calls from Site B to establish a (H323 and SIP) work well, but calls from Site A to Site B work only for H323. Strange things here is localisation tools show the address cannot be resolved and true for this research nevertheless calls SIP of the Site A to Site B do not work when calls is launched of a Cisco endpoint. I already spent a few days and double checked everything but can't break this case, someone knows what's the problem?

  Thanks in advance. Mark

  Hi Mark,

  Since you are able to make SIP calls to internet via MCU and the zones are configured as donotcheckcredential, I think that the problem is with your endpoints rather than VCS. try this:

  -check the DNS configuration on your terminals (EX-series, etc.) and make sure that is correct and the connectivity is here

  -Make sure that the outgoing setting is set to off

  -Make sure the VCS FULL domain name are correct and not poorly spelled and solvable

  s ' there is no proxy between VCS and endpoints server, make sure that this field is empty.

  -I don't think issues of NTP server for SIP calls, make sure however NTP server is configured and works for endpoints

  -one last thing, try saving the end point that is already registered for site A to site B and try to make a call internet SIP, would this work?

  Best regards, Ahmad

• TC7.2.1 problems with SIP calls

  Hi, we had the following problem with sip calls:

  Here is the log:

  Version of the VCS software: Platform X7.2.3 | X8.2.1 Labor

  ______________________________________________________________________________

  Call with TC 7.2.1 does not
  ______________________________________________________________________________

  SIPMSG:
  | INVITE sip: [email protected] / * / SIP/2.0
  Via: SIP/2.0/TCP xx.xxx.60.133:57597; branch = z9hG4bK38f566387dbd719702049ddcb2590ebe.1; rport
  Call ID: 4832bd6a60d9d3d43670c55c573667ec
  CSeq: INVITE 101
  Contact: sip: [email protected] / * /: 57597; gr = urn: uuid:6 c 856033-4cbd - 5 b 15 - bb13 - 0715c4a694cd, ob>
  From: "201101102_C40_ServiceDesk" sip: [email protected] / * />; tag = d25377890df67530
  To: sip: [email protected] / * />
  Max-Forwards: 70
  Directions: sip:xx.xxx.1.51; lr>
  Allow: INVITE, ACK, CANCEL, BYE, update, INFO, OPTIONS, CONSULT, INFORM
  User-Agent: TANDBERG/521 (TC7.2.1.cb31c3d)
  "Proxy-Authorization: Digest ="bb1d830b2d64949caaadfb732cd7ca77414a991699854e8971970a68fe06"nonce, realm ="tplabvcsc01.xyz.com", qop = auth, opaque = 'AwAAAMSaxlh37 + YNQULdXHDdMkXYHVQ1', user name =" ", uri ="sip:xyz.com", answer is"51f225da16a3ac710e86c1b1b5815438", algorithm = MD5, nc = 00000008 cnonce ="21c839c8313e8f7f7477786cb40c5ee6. "
  Supported: replaces, timer, 100rel, gruu, way, way out
  Session time-out: 1800
  Content-Type: application/sdp
  Content-Length: 2842
   
  v = 0
  o = xx.xxx.60.133 IN IP4 6 2 tandberg
  s = -.
  c = IN IP4 xx.xxx.60.133
  b = AS:6000
  t = 0 0
  m = audio 16424 RTP/AVP 107 108 109 110 104 105 9 15 18 8 0 101
  b = TIAS: 128000
  a = rtpmap:107 m4as-LATM/90000
  a = fmtp:107 profile-level-id = 25; object = 23; bitrate = 128000
  a = rtpmap:108 m4as-LATM/90000
  a = fmtp:108 profile-level-id = 24; object = 23; bitrate = 64000
  a = rtpmap:109 m4as-LATM/90000
  a = fmtp:109 profile-level-id = 24; object = 23; bitrate = 56000
  a = rtpmap:110 m4as-LATM/90000
  a = fmtp:110 profile-level-id = 24; object = 23; bitrate = 48000
  a = rtpmap:104 G7221/16000
  a bitrate = 32000 fmtp:104 =
  a = rtpmap:105 G7221/16000
  a = fmtp:105 bitrate = 24000
  a = rtpmap:9 G722/8000
  a = rtpmap:15 G728/8000
  a G729/8000 rtpmap:18 =
  a = annex b fmtp:18 = yes
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  m = video 16426 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:126
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 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
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 11
  a = answer: full
  a = content: hand
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = application 28300 UDP/BFCP
  a = the installer: actpass
  a = confid:1
  a = userid:6
  a = floorid:2 mstrm:12
  a = floorctrl:c - s
  a = connection: new
  m = video 16428 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:126

  
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 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
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 12
  a = content: slides
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = 16430 RTP/AVP 100 application
  a = rtpmap:100 H224/4800
  a = sendrecv
  m = application 29789 UDP/UDT/IX
  a = ixmap:0 ping
  a = ixmap:2 xccp
  |
  ________________________________________________________________________________________________

  Work of appeal with TC 7.1.4
  ________________________________________________________________________________________________

  SIPMSG:
  | INVITE sip: [email protected] / * / SIP/2.0
  Via: SIP/2.0/TCP xx.xxx.60.133:37491; branch = z9hG4bKe3dfed840a299516919a79e4a54cd707.1; rport
  Call ID: 480ec504782c0ad894929e882a696e74
  CSeq: INVITE 100
  Contact: sip: [email protected] / * /; gr = urn: uuid:6 c 856033-4cbd - 5 b 15 - bb13 - 0715c4a694cd, ob>
  From: "201101102_C40_ServiceDesk" sip: [email protected] / * />; tag = 3ab87b1c0d8b4d9d
  To: sip: [email protected] / * />
  Max-Forwards: 70
  Directions: sip:xx.xxx.1.51; lr>
  Allow: INVITE, ACK, CANCEL, BYE, update, INFO, OPTIONS, CONSULT, INFORM
  User-Agent: TANDBERG/520 (TC7.1.4.908e4a9)
  Supported: replaces, timer, 100rel, gruu, way, way out
  Session time-out: 1800
  Content-Type: application/sdp
  Content-Length: 2842
   
  v = 0
  o = xx.xxx.60.133 IN IP4 3 2 tandberg
  s = -.
  c = IN IP4 xx.xxx.60.133
  b = AS:6000
  t = 0 0
  m = audio 16394 RTP/AVP 107 108 109 110 104 105 9 15 18 8 0 101
  b = TIAS: 128000
  a = rtpmap:107 m4as-LATM/90000
  a = fmtp:107 profile-level-id = 25; object = 23; bitrate = 128000
  a = rtpmap:108 m4as-LATM/90000
  a = fmtp:108 profile-level-id = 24; object = 23; bitrate = 64000
  a = rtpmap:109 m4as-LATM/90000
  a = fmtp:109 profile-level-id = 24; object = 23; bitrate = 56000
  a = rtpmap:110 m4as-LATM/90000
  a = fmtp:110 profile-level-id = 24; object = 23; bitrate = 48000
  a = rtpmap:104 G7221/16000
  a bitrate = 32000 fmtp:104 =
  a = rtpmap:105 G7221/16000
  a = fmtp:105 bitrate = 24000
  a = rtpmap:9 G722/8000
  a = rtpmap:15 G728/8000
  a G729/8000 rtpmap:18 =
  a = annex b fmtp:18 = yes
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  m = video 16396 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=245000;max-fs=9000;max-smbps=245000;max-fps=6000;max-rcmd-nalu-size=3456000 fmtp:126
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 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
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 11
  a = answer: full
  a = content: hand
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = application 29489 UDP/BFCP
  a = the installer: actpass
  a = confid:1
  a = userid:3
  a = floorid:2 mstrm:12
  a = floorctrl:c - s
  a = connection: new
  m = video 16398 RTP / AVP 97 126 96 34 31
  b = TIAS: 6000000
  a = rtpmap:97 H264/90000

  
  a = packetization-mode=0;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:97
  a = rtpmap:126 H264/90000
  a = packetization-mode=1;profile-level-id=428016;max-br=5000;max-mbps=108000;max-fs=3840;max-smbps=108000;max-fps=6000;max-rcmd-nalu-size=1474560 fmtp:126
  a = rtpmap:96 H263-1998/90000
  a = fmtp:96 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
  a = rtpmap:34 H263/90000
  a = fmtp:34 cif4 = 1; CIF = 1; QCIF = 1; maxbr = 20000
  a = rtpmap:31 H261/90000
  a cif fmtp:31 = = 1; QCIF = 1; maxbr = 20000
  a = label: 12
  a = content: slides
  a = rtcp-fb: * nack fold
  a = rtcp-fb: * ccm FIR
  a = rtcp-fb: * ccm tmmbr
  a = sendrecv
  m = 16400 RTP/AVP 100 application
  a = rtpmap:100 H224/4800
  a = sendrecv
  m = application 31127 UDP/UDT/IX
  a = ixmap:0 ping
  a = ixmap:2 xccp

  Another problem, indicated in diagnosis on a MX200 device tests, reporting problems of NTP. I tried to choose another NTP server - but this does not solve the problem, and it works with TC7.1.4. In addition to the time and date is wrong on the screen but OK on the touchscreen device.

  In addition, an error is reported on the OSD settings.

  Do you have advice?

  Thanks for help.
  |

  So, what's the problem? You just showed us an INVITATION. What is the configuration of your system? What is the reaction to INVITE him? We need complete diagnostic logs. Is this VCS - C registred? Where VCS - C diagnostic logs

• problem of presentation on SIP calls in CUCM-VSC

  Hi all

  If you would notice, I will be happy.

  Where an end point is registered to CUCM - 8.6.2.21900 - 5 gave a lecture on the MCU MSE 8510, sharing presentation is shown on the media channel, it uses no content channels. But the same endpoint is registered in the VCS - X.7.2.1 and to appeal to MCU SIP, there is no problem, the channels of media and presentation are different.

  End point SIP call-->--> VCS--> MCU CUCM - media and presentation are on the same channel.

  Call--> VCS--> MCU SIP end point - media and presentation are on different channels.

  Thank you.

  Hi Onur,

  Take a look at the following document-

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/VCs/config_guide/Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_8_9_and_X7-2.PDF

  Page 46 shows you what you need to change to get the BFCP working between CUCM and VCS

  Thank you

  Guy

• VCS or highway: force not encrypted video sip call

  Hello community support.

  Let's say I have a vcs cisco or cisco Expressway server and I want to make a call video sip.

  to facilitate the resolution of problems, I want to have a sip call unencrypted, even if the other side external is capable of encryption (has a _sips SRV record).

  is it possible to do on X8.6.1?

  what I've tried so this withour success: (I tried in this order)

  -> Expressway-Edge: defining the Protocol of emergency (B2B) DNS Zone to the mode of encryption TCP transport and the media to "force unencrypted.

  -> the two highways: "TraversalClient (B2B) SIP" area setting transport encryption mode 'TCP' and the media to "force unencrypted.

  -> CUCM isn't in mixed mode

  ->, but the call is always encrypted Highway to the external client.

  -> When I put "Configuration-> protocols-> SIP-> TLS mode on off the call is aborted and no signal is sent to the other side."

  Based on your previous post:

  "with the outgoing call I mean it's a B2B call of an endpoint is saved to CUCM and then uses the course the way Express series server firewall to connect to a SIP Client on the outside." (external domain) »

  On my understanding, it is a vocation of B2B ARM deployment.

  CUCM controlled environment with ARM in which any calls to or from the highway E, which uses the crossing area of CUCM-Expressway is always encrypted. CUCM (X8.6 Highway) crossing area is automatically configured with the proper settings and cannot be changed which have been set to SIP TLS with TLS check the mode of commissioning and mode of encryption Media Force encryptedvalue.

  Kind regards

  Acevirgil

• Use the command line to call EX60/C40

  Hello

  I'm looking to use the command line or xml to call extensions of our EX60 and C40. I want to automate the call made by the devices. We use 9.1 CM and TC6.

  In addition, if someone knows if it is possible to configure ERA/Hot Dial so when a user picks up the phone on the EX60 it automatically connect an extension? I tried through dialing SIP and ERA rule in CUCM.

  Thank you

  Kaleb

  To dial the command line, you can use the command 'xcommand dial.

  xCommand Dial

  Dial a number from the system. Returns information on the CallId and ConferenceId, which are necessary for some other commands.

  Number: Enter the number or address.

  Protocol: Select the Protocol h.323 or SIP.

  CallRate: Fix a rate of call.

  CallType: Select the type of audio or video call.

  BookingId: any identifier that an external booking system (e.g., TMS, CTS - MAN) can use for its own references to match calls placed with the internal identifier systems of booking for a meeting. This can be any string, for example, a GUID. The reservation Id will be provided in the call connects, events of appeal etc. for the call.

  Requires the user role: USER

  Parameters:

  Number (r):
  Protocol:
  CallRate:<64..6000>
  CallType:
  BookingId:

  Example:

  xCommand Dial number: 543210 Protocol: h323
  Ok
  * r DialResult (status = OK):
  CallId: 2
  ConferenceId: 1
  * end

• Linksys E4200 and VoipBuster sip call

  Hello

  Just recently bought a router Linksys E4200 and it was rocking so far! I've updated since a WRT320N. However it seems that the call SIP (using my account Voip Buster) no longer works...

  I don't seem to be able to register with the Voip service. Port is 5060 standard (if I remember correctly, I'm not at home now) and I've already enabled on the router the special tick for Voip calling ALG. Router is upgraded to the latest firmware and it's the V1 router.

  With all the foregoing, I cannot yet connect to VoipBuster using SIP. It worked like a charm with the WRT320N, so it's something with the router.

  Can anyone help? Thanks a bunch!

  Try the following steps:

  [A] with the help of the Cisco Connect software (if at all, you have installed the router using this software)

  1] open the software and go to the option that says "settings of the router.

  [2] then click on the option that says 'Advanced settings', which will take you to the router configuration page.

  [3] lower the MTU (Maximum Transmission Unit) of 1500 to 1400 or less (usually found on the main page / basic configuration of your router)

  4] click administration and disable UPnP.

  [B] If you have not installed Cisco connect then you can connect to the router's user interface using its default IP address in the browser which is 192.168.1.1 and enter "admin" as password empty username field. This will take to the web interface of the router and then follow the steps as indicated above to make the changes.

  Second, it also depends on the type of VOIP provider, you must open the ports on your router. H.323 uses 1719 and 1720 for signaling, dynamic ports for media. SIP tends to use 5060 for signage, dynamic again for the media.

  Follow this link and open the ports on your router to your VOIP: http://www6.nohold.net/Cisco2/ukp.aspx?pid=80&login=1&app=search&vw=1&articleid=17241

• Disable the message/call Cisco C40 failed alert

  Hello

  We have an installer of the C40 codec with control system. The touch screen has very limited control over the video-conference system, so not able to reject the "missed calls" message or alert that appear on the screen.

  The missed call alert appears permanently on the screen, even after you have launched another call. Is there a command API or how to turn off the prompt to alert/message missed calls?

  We cannot disable the OSD. It is one of the requirements.

  Hello

  In the next version of TC6, there will be a command that will disable the notification shown on the OSD display (if you do not use the key).

  xConfiguration video OSD MissedCallsNotification:

• SIP call failures

  Hey all,.

  I'm an EX90 out in the field, running TC6.3.  The EX90 is registered via H323 & SIP at our VCS-E that is running X7.2.2.  The user can call our via H323 MCU without problem, but if the same call via SIP of his unit it disconnects immediately and says "No multimedia codecs shared" in the VCS.

  I looked at all the configs on his unit, and they are fine.  I'm not able to reproduce the problem from the other units.

  EX90 joint papers and they have been taken with the 'Extended loggin' enabled.

  Thank you

  Justin Ferello
  Technical Support Specialist
  KBZ, a Cisco authorized dealer
  http://www.KBZ.com
  e/v: [email protected] / * /

  There is no shared multimedia codecs from the MCU - it supports encryption, as endpoint is dealing with media encryption forced - RTP/BIBI, if xConfiguration conference 1 encryption Mode is set to BestEffort, so maybe it's the profile type parameter that requires.

  You will likely see the MCU logs to see what he doesn't like.

  Thank you

  Guy

• Dialing SIP calls SIP Trunk (Avaya SM)

  Hello. I work with the telco team to send sip of our endpoints VTC voice calls to an Avaya Session Manager.

  I think I have all the areas and rules of research on the side of VCS in order to transfer the call to the SM.  I am able to confirm with the team of the phone company that the call reached the SM, but they receive the error "cannot determine realm."

  On the side of the VCS, I see the output below.  Is the problem on the VCS or SM side?  I need to create a trust with the SM? THX. - F

  • searchRule (2)
  • • Name: SIP Audio out rule2
  • • Zone (1)
    • • Name: TEST Session Mgr
    • • Type: neighbor
    • • Protocol: SIP
    • • Found: false
    • • Reason: Internal server error (cannot determine the domain)
    • • StartTime: 2015-03-09 13:21:58
    • • Time: 0.05
    • • Access controller (1)
      • • Address: 172.x.x.x:5061
      • • Alias (1)
        • • Type: Url
        • • Origin: unknown
        • • Value: [email protected] / * /

  Another option is to see if you get up with 5060/udp or 5060/tcp, which could make it easier to trace. But yes, I prefer the encryption as well.

  Often you have set a kind of trust for a specific connection, which could include the field src or dst, ip, port,...

  good luck ;-)

• Questions call non-Traversal

  Hi all

  I implement some new devices of telepresence:
  Site has
  VCS-control (X8.1)
  VCS-Expressway (X8.1)
  MCU 5310 (4.4)
  C40 codec (7.0.2)

  Site B
  (7.0.2) codec C20
  The site has different subnet

  I have enabled H.323 & SIP on the VCS - C and tested several calls between Site A & b. whenever calls go out H.323, SIP, or SIP to H.323 from the C40 for the C20 or vice versa, that there is no problem. Both sites receive video and audio.
  Whenever I force the same protocol. That is to say, H.323 to SIP for SIP or H.323 and making the same calls, the site B (C20) is not video of the Site (C40). However the C40 receives both video & audio of the C20. What is interesting, is that if I do the same calls of the MCU, which is also in Site A, then there is no problem. So the problem is between the C40 and C20.
  Since I'm not physcally on sites because I do this execution remotely, I can count only on the stats I get from the GUI. I have attached a photo taken of the customer from the screen of the C20.

  Calls:
  Calls crossing
  C40 - C20 H.323 - SIP - call ok
  C20 - C40 H.323 - SIP - call ok
  C40 - C20 SIP - H.323 - call ok
  C20 - C40 SIP - H.323 - call ok
  MCU - C20 H.323 - H.323 - call ok

  Non-Traversal calls
  C40 - C20 H.323 - H.323 - no video received the C20
  C40 - C20 SIP - SIP - no video received the C20
  C20 - C40 H.323 - H.323 - no video received the C20
  C20 - C40 SIP - SIP - no video received the C20

  This suggests that there is only a problem on the C20 in non-traversal calls. I don't want to use licensed traveled between these 2 points of termination.

  Thank you for your help. Any input would be greatly appreciated.

  -Greg

  When you make an H323, SIP call, all media is 'mandated' by the VCS, so they would still work even if they cannot ping each other.  When you do a non-traversal call, media flows directly between the endpoints.

  Ping a C-Series, connect to the codec via SSH (same credentials as the Web browser), and then type systemtools network ping 10.1.1.1

  Where "10.1.1.1" is the IP address of the remote end point.

• Source IP address of the originating call to sip trunk

  I'm unable to set address ip source from trunk sip call. I read somewhere that the source ip address of the call must be the ip address that call device registered to. I also checked "run in all active nodes such as suggested by cisco. We have 3 Sub cluster and 1 pub however still call orgianate to a particular Sub.

  I also undergoes several changes i - e DP and list route but think not worked for me. Can someone help to sortout this problem?

  Let's say if your ip phone registered in sub - and "run on all nodes" enabled, then the source ip address is always sup - A s ip address regardless of the device pool config on the sip trunk.

  Could please explain you the device pool config in phones IP & SIP trunk?

  you have enabled the "run on all nodes" for a list of course also?

  Suresh

  Please note all useful messages.

• How to replace the call number part on a SIP section if none is present

  Hello

  I have a 2821 router that is connected to a SIP trunk.  This router also has a PRI interface that is connected to a PBX Mitel.  The Mitel used the router has a primary connection to the PSTN.

  The Mitel will send the party call number most of the time, but sometimes it isn't.  All calls that are sent to the SIP provider without a party call number are deleted.  I don't know if can substitute part number calling using the command in clid on the Dáil peer, however is it possible only override the calling party number if I do not get one of the Mitel?  He got a number of appeal left like a DID I want to pass along in the SIP call.

  Thank you.

  You should be able to do it with a voice translation rule that matches an empty ANI.
  30rule translation-rule voice 1 / ^ $\+631689\(...\) / / 01689\1/rule 2 / ^ $/ /555666777/
  translation-profile PSTN-OUTtranslate of the voice calling 30

  First rule to match a real number to a PBX, to adapt according to your needs. The second rule matches a null value and changes to 555666777, who adapt to what you would like to send to your phone company.

  Do not forget to rate helpful responses and identify useful or correct answers.

• SIP spam attack and MCU and vcs - e call

  as far as I know sip call spam attacks is done against the videoconference, connected with a public ip address, I disabled the sip but im not sure if my mcu and vcs - e with sound are vulnerable to them? they pose no threat to security for them? and if so, how? and what can we do about it?

  It is a well known problem and it affects H.323 and SIP, take a look at the below threads:

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

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

  https://supportforums.Cisco.com/discussion/12508641/Cisco-source-spam-calls-stepped-complexity

  https://supportforums.Cisco.com/discussion/12613681/attack-vcse

  There are many more discussions on this issue, the above, this is just a small selection. :)

  You do not need to disable SIP on the VCS-E, all you need to do is turn SIP UDP unless you need it for voice services.

  You can protect yourself by using a CPL on the VCS-E who will avoid calls to go through your MCU, or anything else you have sitting behind the VCS-E. This is assuming that you are using a combo of VCS-C/VCS-E, with the VCS - C behind a firewall and the VCS-E outside the firewall, for example in the demilitarized zone.

  Having just trouble ask points of termination or MCU sitting in nature with public IP addresses.

  These scans, moreover, mainly looking for systems that will allow them to make free international calls.

  /Jens

  Please evaluate the answers and makr as 'answered' questions as appropriate.