Linksys E4200 and VoipBuster sip call

Similar Questions

• Linksys E4200 and connectivity Motorola SBG6580

  Here is a picture of my home network. The problem I'm having is, I can't get the WD MyBook Live Duo to be seen on the entire network. If it is connected to the Linksys, all computers connected to the Linksys can see it, but not those related to the Motorola. If I put it on the Motorola, the same thing happens, but in the visa versa. I bought the Motorola for its DOCSIS 3 capabilities, not for the wireless. I prefer to use the Linksys Wireless. Any help would be greatly appreciated.

  

  Devices must be on the same subnet (e.g. 192.168.1.x) to be able to see each other. You could make the Sb IP 192.168.1.1 and 192.168.1.2 for the E4200 and connect a cable from LAN of the SB E4200 (not WAN) port lan port.

  Here is a guide if you need: http://www6.nohold.net/Cisco2/ukp.aspx?pid=80&vw=1&articleid=3733

• V - Portal of Vonage and Linksys RTP300 screw up the call diversion

  I have a situation where a Linksys RTP300 legacy and vestigial is unconnected to all phones, but only to the LAN seems to screw up the call between Vonage numbers forwarding phones attached to a single Vonage V-portal which can be found on the same LAN.  Incoming and outgoing calls are not affected. Remove the Linksys RTP300 seems to fix the problem.  Can someone tell me why this would happen because the RTP300 is not connected either for all phones?  And why the call forwarding is only affected (tent to transfer calls sends just calls to your voicemail, even if the other line is not picked up or not in use).  The network is flat where the V-Portal of Vonage Linksys RTP300 and one small switch 8 ports are linked directly to the five ports integrated switch a firewall router, with the WAN port on the firewall router attached to a T1 CSU/DSU.  I'm assuming (perhaps wrongly), if no phone is connected to the RTP300, there would be no SIP traffic either.  Or would the RTP300 send SIP traffic even with no attached phone?

  Hello.

  These products are processed by the Cisco Small Business Support Community.

  * If my post answered your question, please mark it as "acceptable Solution".

  Thank you!

• Linksys E4200 - can update firmware and roll backs damage my router?

  Hello

  I had to restore the firmware on my Linksys E4200 (v2) because I had problems after upgrading the firmware. I tried to upgrade again and again met with problems. In short, I improved a few times and restored several times. Currently I'm on the old firmware and I without any problem.

  In the old days, it was not wise to update the firmware of some peripheral hardware too often, because some devices has agreed that a limited number of updates to firmware... don't ask me why :-)

  Is this also the case with the Linksys E4200 or can I upgrade and roll back as often as the requires, or as often as I want to try to upgrade because of new features...?

  Thank you

  SJW

  As long as you do the right steps during the upgrade/downgrade the router (loading the firmware using a wired computer, not to interrupt the upgrade/downgrade process, load the right file, etc.), the unit should be fine.

• 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.

• Linksys 4200 and Samsung Galaxy Tab 10.1 with 5 GHz

  Hello!

  I just bought Samsung Galaxy Tab 10.1 wifi.
  I have the router Cisco Linksys E4200.

  I have activate 5 GHz on E4200.

  I have disable 2.4 GHz on E4200 to be sure that the tablet will connect at 5 GHz.

  But, in doing so, the Tablet does not have wifi.

  Someone managed to connect at 5 GHz with this router and this Tablet?

  Thank you very much!

  Set the 5 GHz band on the router to a single, 20 MHz, and channel 36.

  Search for again the 5 GHz SSID.

  I guess it comes tab or the router and a band 5 GHz default radio.

  You have another dual-band device to test?

  Or the tab shows which group is used for other wireless networks around? Or maybe a network scanner? If she find any access points 5 GHz...

• Linksys E4200 - Skype internet killing

  Hello

  Recently, I went from a router D-Link GamerLounge for a new Linksys E4200 using the default firmware, upgraded to the latest version.

  The router seems to work fine - except when someone uses Skype, and performance of the network decreases drastically. I have tested it with Skype running on a wireless laptop and test my Wired PC network. Here are the results of ping www.google.com with and without Skype:

  Without Skype (wired PC ping):

  Reply from 74.125.113.103: bytes = 32 time = 132ms TTL = 46
  Reply from 74.125.113.103: bytes = 32 time = 83ms TTL = 46
  Reply from 74.125.113.103: bytes = 32 time = 183ms TTL = 46
  Reply from 74.125.113.103: bytes = 32 time = 105ms TTL = 46
  Reply from 74.125.113.103: bytes = 32 time = 48ms TTL = 46

  With Skype (running on the laptop wireless, PC ping wired):

  Reply from 74.125.113.103: bytes = 32 time = 132ms TTL = 46
  Reply from 74.125.113.103: bytes = 32 time = 83ms TTL = 46
  Reply from 74.125.113.103: bytes = 32 time = 183ms TTL = 46
  Reply from 74.125.113.103: bytes = 32 time = 105ms TTL = 46
  Reply from 74.125.113.103: bytes = 32 time = 48ms TTL = 46

  I searched a bit on the internet and have not found anyone with a similar problem. Someone at - it suggestions? I guess I could try to upgrade the firmware again or something.

  Probably due to the quality of service. You said it was a GamerLounge, which I suppose has automatic QoS for the games.

  You can configure QoS on the linksys as well, it's just not the installer for the game by default.

• Redirect with e4200 and 660 embarq modem port

  Hello, I have a router e4200 and a modem 660 series, I am trying to portforward for a game called minecraft, I can't get ports open, I called century link they told me to put my modem in Bridge mode, so I have, they said, must be public, my private IP not now, it is still private, what I am doing wrong.

  Very well. Great! Now follow these steps and see what happens.

  http://www6.nohold.NET/Cisco2/UKP.aspx?pid=93&login=1&VW=1&app=search&articleid=17241&userrole=links...

  In addition, under the tab 'security' on the router, uncheck the option, "filter anonymous Internet requests".

  The tab of the router configuration. There is an option 'MTU '. Change it to manual and reduce the size to 1400.

  See if that helps.

• Linksys E4200 - range WUSB600N V2 / signal issues

  Router Linksys E4200 dualband and Win7 64 bit Linksys WUSB600N v2.
  Both in the same room, the distance is about 10 meters.
  I compare the results on 5 GHz btw the stick of Linksys and btw another laptop with Intel Advanced-N 6200 AGN wifi chip.
  Running a wifi 5 GHz N on the stick of Linksys, leading to the signal strength low and reduced its speed.
  Practically, I can't improve the signal than the Intel + 15 to 20 dB...

  Negligible:

  distance / Intel / Linksys:
  <1 m ="" -26db="">

  5 m /-50 dB /-65dB

  While Intel radio affects 300 MB/s, the Linksys is somewhere around 200 MB/s only. in the same room
  Anyone else knows a bad signal with the WUSB600N v2, as well? I bought the stick to stream HD media, but with - 65dB is not much of an improvement to what I had on 2. 4 Ghz, 802.11 g (())

  Sorry, but how is this releated response must keep the problem I described?
  disable the firewall on the router spi?
  I checked the possibility of automatic channel selection and channel 20-40 establishments width (automatic channel selection does NOT work correctly from either). I still have the experience that the stick has lower than intel one signal. regardless of the setting of the router firewall 

• E4200 and Netgear WN802T-200 AP Configuration?

  Hello

  I plan to use the Linksys E4200 as the main router downstairs for wired and wireless - n 2.4 and 5 GHZ access. Upstairs, I have a Netgear WN802T-200 wireless-N Access Point uses 2.4 wireless only. What is the best way to set up the network?

  My House is wired CAT5, after I connect the Netgear to the E4200 access point. Set the two devices with the same SSID and wireless channel?

  It is the best practice, or is there another method of setting up the network for better performance and the beach?

  Thank you

  Sounds good, will keep them on auto channel.

  Thanks fir the quick response.

• Courses and non-traversal calls

  Hello

  Can someone explain in simple terms what are courses and non-traversal calls?, what is the difference? And also the VCS control below must traversal licenses?

  CTI-VCS-CONTRL-K9

  Control of VCS

  Thanks in advance.

  If you have two endpoint C-series (i.e. CTS-SX20) both recorded on VCS control and also capable of reaching out directly between them (a routing), then in general the call will address as non-traversal call Endpoint, so no license courses will be necessary.

  However if these Endpoint calling each other, but using different protocol (i.e. an end point registered in VCS control only with H323 and other endpoint registered on the CV with SIP), or different IP version (that is to say one with IPv6 and other with IPv4 registration), then control VCS remain media routing in order to manage such scenario of interoperability which will require licensed traversal appeal.

• 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

• Rogue SIP calls on C40

  Hello

  We have a videoconference system with the public IP address that is registered in the VCS-E received various rogue SIP calls.  All these SIP calls are aliases for figures 3-6 during 32sec each.  The calling address entering is: alias @ public IP address of our VC system. All calls are video calls to 384 k

  To avoid calls, I disabled the SIP option and since then we have not received one of these calls.  However, we need to connect to other devices using SIP.  Is there another way to stop these calls?

  Thanks in advance for your help.

  Amrit

  Make the following settings on your endpoints:

  • xConfiguration SIP ListenPort: Off
  • xConfiguration SIP profile 1 out: on

  See bug CSCue55239 for more details.

  You will also need to take steps to secure your VCS if you have not already, turning off UDP SIP SIP calls stop.  However, last year, we saw these calls come on H323 TCP, the only way to stop calls H323 is either secure endpoint behind your firewall and use a script CPL on the VCS.  See the analysis of sourceh323idcisco-incomingcalls on how to configure a CPL.

  FYI, the forums to search would be my first place to look, or search for bugs.  He is asked everywhere in the forums.

• 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