SIP call failures

Similar Questions

• 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

• 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

• 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

• 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 ;-)

• BlackBerry Smartphones blackberry 8900 curve: call failure needed help

  Hello.

  I have a curve 8900 that I got off of ebay from a seller who got very good food back. However the when I got on the phone he has an orange logo on the back of the phone. the seller has unlocked the phone. but everytime I make a call from the phone it says "call failed". Please can someone help as I want to get the phone working again, if possible.

  Please help because it's urgent.

• BlackBerry smartphones [! call failure]-cannot make or receive calls/texts incoming/outgoing but internet/email still works fine?

  I just bought a replacement bb bold 9000 off ebay (it's an at & t unlocked handset I bought after my bb bold got stolen... total disappointment) and it was working fine all week until last night when he stopped being able to manage incoming and outgoing calls and texts. E-mail and internet functions still work that is very special.

  also, some more info, when I put my sim card into another handset (in this case a nokia e65) all my work of normal calling features and when I put a different SIM in the "BOLD" the same thing happens (when I try to make a call, it instantly opens with a warning of the call). Another thing is that the "BOLD" indicates full service coverage...

  currently the "BOLD" on the fairs...

  v4.6.0.304 (Platform 4.0.0.247)

  Note: im on the Australia and for some reason, I can't update the os v5

  Here, any help would be greatly appreciated, and I would be happy to provide any information to anyone who is willing to help.

  jenm_ wrote:

  apparently v5.0 is still not available on my carrier / in my country (aus)) and yet its does not work...

  any advice would be so greatly appreciated!

  There you go. This is the reason.

  For the missed show called, I don't know. The unit was unlocked when you bought on eBay?

  My first suggestion would be do not have the feature of your problem. Return it and get your money back.

• 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.

• 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 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.

• Translate the name of SIP - UA number called

  Did anyone know, how to use the number called in SIP calls instead of the name of register SIP - UA.

  In my case A7302337 - authentication name, 7302338 - incoming called number.

  But, dial-peer unknown why use of A7302337.

  See below.

  Thanks in advance!

  C2901. Lawer #.

  C2901. Lawer #.

  C2901. Lawer #.

  C2901. Lawer #.

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

  Received:

  GUEST sip:[email protected]/ * /: SIP-5060/2.0

  Via: SIP/2.0/UDP 89.207.94.24:5060; branch = z9hG4bK464550; rport

  P-CGP-redirector: [email protected] / * /

  Record-Route:

  Record-Route:

  Via: SIP/2.0/UDP 89.207.94.25; branch = z9hG4bKac2045160544

  Max-Forwards: 69

  From: <> [email protected]/ * / >; tag = 1 c 2045153887

  To: <> [email protected]/ * /; user = phone >

  Call ID: [email protected]/ * /.

  Contact: <> [email protected]/ * / >

  CSeq: 1 INVITE

  Support: em, 100rel, timer, replaces, path, start of session, resource-priority

  Enable: REGISTER, OPTIONS, GUEST, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, REGISTER, update

  User-Agent: Audiocodes-Sip-gateway-Mediant 1000/v.5.20A.021.001

  Content-Type: application/sdp

  Content-Disposition: session

  Content-Length: 569

  v = 0

  o = AudiocodesGW 2045145418 2045145099 IN IP4 89.207.94.25

  s = phone calls

  c = IN IP4 89.207.94.25

  t = 0 0

  m = audio 6770 RTP / AVP 0 8 18 101

  c = IN IP4 89.207.94.25

  a = rtpmap:8 PCMA/8000

  a = rtpmap:0 PCMU/8000

  a G729/8000 rtpmap:18 =

  a = fmtp:18 annex b = No.

  a rtpmap:101 telephone-event/8000 =

  a = fmtp:101 0-15

  a = sendrecv

  a = ptime:20

  a = rtcp:6771 IN IP4 89.207.94.25

  m = image 6772 udptl t38

  c = IN IP4 89.207.94.25

  a = T38FaxMaxBuffer:1024

  a = T38FaxMaxDatagram:122

  a = T38FaxRateManagement:transferredTCF

  a = T38FaxUdpEC:t38UDPRedundancy

  a = T38FaxVersion:0

  a = T38MaxBitRate:14400

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number = A7302337, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled is 1 saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = 89036223360, called number =, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = 89036223360, called number =, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpAssociateIncomingPeerCore:

  Number = 89036223360, called number = A7302337, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/580225688208/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled = 0, saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled = 0, saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number = A7302337, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled = 0, saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = A7302337, called number =, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Number = A7302337, called number =, Voice-Interface = 0 x 0.

  Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,

  Peer Type Info = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

  Result = NO_MATCH(-1) after all rules attempt Match

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled = 0, saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchSafModulePlugin:

  dialstring = A7302337, saf_enabled is 1 saf_dndb_lookup = 1, dp_result =-1

  May 17, 14:06:34: //-1/580225688208/DPM/dpMatchPeersMoreArg:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 101, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 101

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40002

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 103, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 103

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40006

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 104, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 104

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40004

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 105, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 105

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40003

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 106, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 106

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40001

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = A7302337, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = A7302337

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  No outbound dial-peer does not; Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = NO_MATCH(-1)

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Number =, called number = 107, Peer Info Type = DIALPEER_INFO_SPEECH

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Associate the rule of = DP_MATCH_DEST; Called number = 107

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

  Result = Success (0) after DP_MATCH_DEST

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:

  dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

  Result = Success (0)

  The outgoing dial matched host list:

  1: Tag dial-peer = 40005

  May 17, 14:06:34: / / 206/580225688208/SIP/Msg/ccsipDisplayMsg:

  Envoy:

  SIP/2.0 100 trying

  Via: SIP/2.0/UDP 89.207.94.24:5060;branch=z9hG4bK464550;rport,SIP/2.0/UDP 89.207.94.25; branch = z9hG4bKac2045160544

  From: <> [email protected]/ * / >; tag = 1 c 2045153887

  To: <> [email protected]/ * /; user = phone >

  Date: Thu, 17 may 2012 14:06:34 GMT

  Call ID: [email protected]/ * /.

  CSeq: 1 INVITE

  Allow-events: telephone-event

  Server: Cisco-SIPGateway/IOS-15.2.3.T

  Content-Length: 0

  May 17, 14:06:34: / / 206/580225688208/SIP/Msg/ccsipDisplayMsg:

  Envoy:

  SIP/2.0 404 not found

  Via: SIP/2.0/UDP 89.207.94.24:5060;branch=z9hG4bK464550;rport,SIP/2.0/UDP 89.207.94.25; branch = z9hG4bKac2045160544

  From: <> [email protected]/ * / >; tag = 1 c 2045153887

  To: <> [email protected]/ * /; user = phone >; tag = 29C6FC-1848

  Date: Thu, 17 may 2012 14:06:34 GMT

  Call ID: [email protected]/ * /.

  CSeq: 1 INVITE

  Allow-events: telephone-event

  Server: Cisco-SIPGateway/IOS-15.2.3.T

  Reason: Q.850; cause = 1

  Content-Length: 0

  May 17, 14:06:34: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

  Received:

  SIP ACK:[email protected]/ * /: SIP-5060/2.0

  P-CGP-redirector: [email protected] / * /

  Via: SIP/2.0/UDP 89.207.94.24:5060; branch = z9hG4bK464550; rport

  Max-Forwards: 69

  From: <> [email protected]/ * / >; tag = 1 c 2045153887

  To: <> [email protected]/ * /; user = phone >; tag = 29C6FC-1848

  Call ID: [email protected]/ * /.

  CSeq: 1 ACK

  Content-Length: 0

  Try to use a voice translation like this rule:

  translation of the voice-rule 1

  rule 1 /A7302337/ /7302338/

  voice translation-profile XLATE_CALLED

  translate 1 called

  If you want to use the To header instead the SIP URI, you must use the CUBE.

  See this article: http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/

  Kind regards.