CUCM v8.5 with 3 SIP Trunks to the Lync Server - Route algorithm of Distribution for the Group

Similar Questions

• is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?

  Dear all,

  Need help :)

  1.

  is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?

  my customers will not buy TP - lic :)

  Topology:

  8900 (jabber) - CUCM-(SIP) - SX10

  2.

  Customer, have MCU machine.

  If not possible. is it possible, meeting point of MCU use?

  Topology:

  8900 (jabber) - CUCM - MCU - SX10

  1. Yes, but it is not that simple and straightforward that the SX10 can be reached by calling is the public IP address and CUCM does not support the IP address numbering is not a SIP address format...

  2. Yes, using the MCU would be much simpler, easier and, in my opinion, a better solution to be implemented. Here is an example of how a stand-alone SX10 could dial the IP address of the MCU standard automatic.

  You can also assign a SIP URI address to the meeting point which the SX10 can connect to.

  /Jens

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

• Where can I find the 2 stucco that comes with photoshop, is not in the group. Co

  Where can I find the 2 stucco that comes with photoshop, is not in the group. Co

  Hello

  The model of stucco 2 should be in patterns of Texture Fill 2 .

  

• Problem with a SIP Trunk between VCS and CUCM

  Hello world

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

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

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

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

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

  You have an idea to solve this problem?

  Thanks for your help

  Best regards

  Bruno Z.

  Hello

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

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

  make G.711 end to end and test again.

  Thank you

  Alok

• Caller ID of VCS Jabber Video on SIP Trunk on the phone

  Hello, Netpros.

  I am trying to find a way to show a caller E.164 ID when a call to a video client Jabber a UCM or a PSTN phone.

  VCSC/E 7.1; 13.2 TMS, configured TMSPE.

  We have a VCSC > SIP Trunk > UCM (8,6) > H.323 GW w / PRI voice configured for the composition of and the PSTN destination.  We do this, so the Jabber client video can be used as a softphone, and so we can have transparent FindMe to voice and SNR to the video in either sense.  In TMS, we have a model of implementation configured with an address of the video, ILA and unit address.  He pulls ad office phone number for video of Jabber users.

  From what I read here:

  http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_FindMe_Express_Deployment_Guide_X7-1.pdf, we should be able to see the caller ID when you call send out an ISDN GW.

  «This section describes how to use FindMe with calls are routed through an ISDN gateway (for example, when you call a mobile phone, or any other destination accessible ISDN).» If VCS has Caller ID (Applications > FindMe > Configuration) defined to use the ID FindMe, the identification of the appellant presented will be the user E.164 phone number. »

  UCM SIP config is vanilla, by

  ttp: / /www.tandberg.com/collateral/documentation/Deployment_Guides/Cisco_VCS_Ci...

  However, when a video call Jabber is directed to a University Complutense of MADRID registered phone, we get "unknown caller".  When to call us directly to the PSTN, we see that the BTN from the PRI.

  What is strange, is that if we call another client of Jabber of VCS (with all the same problems), directly into a phone UCM via Internet on URI (for example [email protected] / * /) through the VCSE, we see the caller ID name.

  Thank you, all!

  Joshua

  Hi Joshua,.

  At present the VCS will only send caller E.164 id to a GW of H.323 ISDN, which is registered for VCS.

  There a feature request in the order book to define if E.164 Caller ID or name/Video address on a per-zone basis. This will also be sove the question when the call is to go on a SIP connection, as in this case, a remote ISDN GW.

  Thank you

  Guy

• SIP trunk CUBE with Callcentric - incoming unanswered call

  I'm doing some tests with a Sip trunk with a provider called Callcentric.
  It is a CUBE scenario. I use a SIP to the CUCM trunk.
  
  I have a Cisco Callmanager and a virtual router 7200 (GNS3) as a gateway (C7200-ADVENTERPRISEK9-M), Version 12.4 (24).

  A CPIC connected to Callmanager, I call out to PSTN and it works perfectly.

  When I do an incoming call to the PSTN to the softphone destination, she sounds, but when I press the answer button, the call is not connected and the analog phone in RTC can listen back ringtone.

  Do you have any idea what it could be?
   

  Some relevant configurations:
   

  voip phone service
  allow sip to sip connections
  Fax protocol cisco
  SIP
  interface FastEthernet0/0 source control binding
  bind media source interface FastEthernet0/0
  Registrar Server

  voice class codec 1
  g711ulaw codec preference 1

  translation of the voice-rule 1
  rule 1 / ^ 8 / /0056/
  !
  voice translation-rule 2
  rule 1 5.0 / /17772114zzz/
  !
  voice translation-rule 3
  rule 1 /17772114zzz/ /500/

   

  voice translation-profile IN
  definition of 3 called
  !
  FLIGHT voice translation-profile
  definition of call 2
  translate 1 called

  Dial-peer voice 1 voip
  CALLCENTRIC description
  entrants IN translation-profile
  translation-profile outgoing OUT
  destination-model 8.T
  codec voice-class 1
  session protocol sipv2
  session target sip-Server
  incoming called-number 17772114zzz
  SIP DTMF-relay-notify rtp - nte
  !
  Dial-peer voice 2 voip
  CUCM description
  destination-model 500
  media stream-autour
  codec voice-class 1
  session protocol sipv2
  session target ipv4:192.168.10.116
  incoming called number 8.T
  SIP DTMF-relay-notify rtp - nte

  SIP - ua
  credentials of username, password 7 17772114zzz 094F471B100928 callcentric.com Kingdom
  authentication username 17772114zzz password 7 121A0C051B0803 callcentric.com Kingdom
  no remote-party-id
  Registrar dns:callcentric.com expires 3600
  DNS:callcentric.com SIP server
  Home-Office

  
  Thank you guys.
   
   

  Hello

  Please configure the sip requires low profile to remove the header and apply this profile in the dial-peer 1 outgoing.

  SIP-class voice profiles 1

  response header 200 sip requires DELETE

  If this does not work under Dial-peers, try to apply globally.

  voip phone service

  SIP

  SIP profiles 1

  Suresh

  Please note all useful posts

• CallManager Express SIP trunk problem

  Hi all here.

  I have a problem with my SIP trunk CallManager Express (version 10.0). On my site already configured trunk SIP between CME and CUE. I have configured DN for SIP phone and SIP phones recorded on CME successfully, but... As soon as SIP phone registered on CME, DIAL-PEER for CME and CUE connection changes. Especially 'session target ipv4' automatically changes to my IP phone SIP and all calls go to my SIP phone and does not reach the auto attendant.

  How can I solve this problem?

  Hello

  Following is the error of CUCM;

  WARNING: 399 UnicompCM "cannot find a Device Manager for the request received on port 55549 leave 192.168.10.2.

  This probably means that CME use 192.168.10.2 for source packages SIP CUCM however SIP trunk in CUCM is not directed to this IP address, it is useful to point out to another IP interface of the GUY and therefore dismiss this appeal.

  Can you please check what is the IP of CMF address you configured on the SIP trunk in CUCM?

  -Vivek

• DTMF in SIP Trunk problem

  Hello

  I have a problem in case of detection of the DTMF

  We have a SIP of the ITSP Trunk and everything is ok except DTMF.

  The sip trunk is between ITSP and router 3945

  ITSP <->3945 <->CUCM 10.5

  I have try all method such as rtp - nte, h245 alphanumeric and h245-signal, info-sip, sip-kpml,... in line-peer to ITSPs

  ITSP say that he send with standard RFC2833 dtmf (it equals to rtp-net, I know), but when I am 'debug ccsip message', the results show the dtmf with rfc2833 sends us

  16 August 13:52:28.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  INVITE sip: [email protected] / * /: 5060. user = phone SIP/2.0
  Via: SIP/2.0/UDP 10.105.40.34:5060; branch = z9hG4bKejhh8jixkobhnb7ykvi87vuj8
  Call ID: [email protected]/ * /.
  From: sip: [email protected] / * />; tag = c3cx3vcw-CC-40
  To: sip: [email protected] / * /; user = phone >
  CSeq: 1 INVITE
  Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see
  Max-Forwards: 69
  Supported: 100rel, timer
  User-Agent: Huawei SoftX3000 V300R010
  Session time-out: 300
  Min - SE: 90
  Contact: sip: [email protected] / * /: 5060; user = phone >
  Content-Length: 374
  Content-Type: application/sdp
  v = 0
  o = HuaweiSoftX3000 11042757 11042757 IN IP4 10.105.40.34
  s = call Sip
  c = IN IP4 10.105.40.34
  t = 0 0
  m = audio 27762 RTP / AVP 8 0 18 4 2 98 99 102
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a G729/8000 rtpmap:18 =
  a = rtpmap:4 G723/8000
  a = rtpmap:2 G726-32/8000
  a = rtpmap:98 G726-40/8000
  a = rtpmap:99 G726-32/8000
  a = rtpmap:102 G726-24/8000
  a = ptime:20
  a = fmtp:18 annex b = No.
  It is a message to guest (with sdp) of ITSP
  As you can see the line with red color must have a code with number of 101 but rather a code with number of 18
  In my "ccsip media debug' output show that the method of negotiation between me and itsp 'incoming voice. '
  It's my router config:
  voip phone service
  No IP trust to authenticate
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  SIP
  interface FastEthernet0/0/1 source control binding
  bind media source interface FastEthernet0/0/1
  min - to 300 session expires-300
  !

  Dial-peer voice 2 voip---> router CUCM and vice versa
  translation-profile outgoing toos
  destination-model 42584...
  session protocol sipv2
  session target ipv4:10.20.30.70
  Codec g711ulaw
  DTMF-relay rtp - nte
  !
  VoIP voice 10 Dial - peer---> router for ITSP and vice versa
  destination-model. T
  session protocol sipv2
  session target ipv4:10.105.40.34
  incoming called-number. T
  DTMF-relay rtp - nte
  Codec g711ulaw
  I have configured cucm with a sip section to my favorite router with active PSG and RFC2833
  BUT HE DIDN'T THERE WAS NO DETECTION OF DTMF IN INCOMING CALLS AND OUTGOING
  I even test dial-position 10 without any method of dtmf-relay because I want to configure it with the incoming voice method, but it does not work
  I change the codec but does not solve the problem
  There is an interesting point and that is, if use Elastix 3945 of the router and configure dtmf among elastix and itsp as "inbound" method and method dtmf among elastix and cucm as 'rfc2833' everything is OK (ITSP<--Inbound--> Elastix <--rfc2833-->CUCM)
  Please give me a solution to solve the problem between Cisco 3945 and ITSP
  Concerning

  It would be more accurate to say that the problem has been bypassed by using a transcoding resource. The correct resolution here would be to open a ticket to trouble with the ITSP. I agree that the PROMPT message you show doesn't include advertising RFC2833 in the SDP offer. It is a problem that only the carrier can rectify.

• SIP trunk behind a router using NAT

  Hello

  Is it possible to use a SIP trunk to a provider SIP ITSP having the CUBE / router gateway behind a firewall using a NAT?

  Does anyone do this?

  I ask because I'm having problems to make my SIP trunk to work and my router for cube is behind my generic service provider router, which makes the NAT. I just want to rule this out as a problem.

  Has anyone else done this? Or is it really impossible?

  Thank you very much

  Tom

  Hello

  As NAT works fine SIP would work properly as the Protocol.

  Here is the RFC for "NAT Traversal practices for Client - Server SIP"

  https://Tools.ietf.org/html/rfc6314

  HTH

  JB

• Configure (fxs) analog phones with Sip trunk

  Hi all

  IS - this configuration FXS possiple to knit on SIP TRUNK? I HAVE 16 FXS ports and voip gateway 2921 cisco.

  Configuring fxs to work with her?

  any help thanks.

  Is there a PBX at stake here, i.e. CUCM or CME?

  In both cases, you have set good dial-peers to point to the FXS ports to match the destination of the SIP trunk, which is pretty simple.

• CUCM SIP-trunk multiple Ports

  I would like to know if it is possible to create a Script of SIP standardization in CUCM which change the Destination SIP port based on the phone which makes the phone.

  Current issue:

  The ITSP provider asked that I send calls from different regions to the same address IP SBC but a different port.

  Example:

  Phone (A) makes a call

  RP - RL - RG---> SIP Trunk (5060)---> CUBE (5060<-->5001)---> ITSP1

  Telephone (B) makes a call

  RP - RL - RG---> SIP Trunk (5060)--->---> CUBE ITSP1 (5060<-->5002)

  I can do it on the CUBE, but which require several dial-Exchange and SIP profiles, is it possible in the CUCM to change the sip by device-pool port?.

  Or is there another I can match and send calls via the same destination with different dial-peer sip ports.?

  Thank you.

  Zakiab,

  You can do this, its all just impossible. Your signage ports are defined not on the endpoints, but on the B2BUA (CUCM or CUBE). The CUBE, you have the most flexibility, because you can change the port of signs based on your dial-peers. However, as you pointed out quite rightly, need you an insurmountable amount of dial-peers to have this for each endpoint. So I suggest tell you your ITSP to change their design or moving to a new.

• secure sip trunk cucm driver

  Hello

  I have set up non-secure sip trunk using tcp of cucm to the driver

  It's up on top of the cucm but in the driver call control is inaccessible

  so the Conference made from an endpoint works but not the scheduled conference or composed auo

  also, I set up a secure safe using tls

  This time sip trunk is down in the side cucm and so it's on the driver side

  I tried to download the temporary driver's certificate in the callmanager-trust and the tomcast-trust

  Download the certificate of call manager to the conductor

  but still does not work

  I don't know if I need to generate a certificate authority certificates or just, I missed something?

  CUCM: 10.5.2

  conductor: XC4.0

  Thank you

  Hello

  I recently deployed the CUCM and conductor (same version) as mentioned above.

  My case was a little different as my certificate CUCM management was made by the internal certification authority, similarly to the conductor, I did the management of certificates, made sure the root certification authority is present in the two server to trust each other.

  So I configured the location with port 5061 with ip address CM on the conductor.

  Similarly on CM SIP trunk pointing to ad-hoc conductor and meeting woth port 5060 and security profile, device security mode an encrypted and the subject name X.509 to match the FQDN or Cluster COMPLETE domain name to match the domain the driver's FULL name, which allows the TLS communication code.

  If I remember I was too faced the same problem when you face, however, after appropriate management of certificates and security profile, it has been resolved properly.

  This guide should be useful

  http://www.Cisco.com/c/dam/en/us/TD/docs/Telepresence/infrastructure/con...

  Please let me know if you need more help.

  Kind regards

  RACLOT

• CUCM 8.5 and "Secondary numbering" with phone SIP

  We have 8.5.1.x the standard Cicso CUCM 'skinny' phones all work very well. The problem is with SIP endpoints on a console to send Avtec. They work fine - BUT once a call is put no more of DTMF tones are sent through the line. I hear the release in the Atrium - but it does nothing at the other end. Example: Put a get call for a "tree of voicemail. Most entries are required to go anywhere. Pressing the button gives a release, but nothing happens. Looking at the Wireshark capture it appears that it is not enabled on the Cisco side (so the Avtec tech guys say). I don't know exactly what Cisco calling - I saw "Secondary numbering" once. What needs to be "activated" in the Call Manager to do. Help! Mark

  It seems that these phones are supported RFC2833. Without a view in your system and you have, it is hard to tell you in the forums where things need to be tweaked.  In this case, the TAC is a good call I think.

  Hailey

• CUCM 8.6 &gt; SIP Trunk &gt; CUBE &gt; SIP Trunk - ringtone / connected display Party

  Hi community support

  When we get calls from an IP phone on a gateway H323 ISDN globalize us the number of models of translation so the display on the phone is the E164 number, for example + 442071234567. We want this even when the call is ringing / connected for us to use the view command no service additional h225 - prevent cid-update on the H323 gateway otherwise the alert screen is updated with the number of the corresponding when Dial peer ringtone and connected.

  When we get calls from an IP on a CUBE phone to SIP the number is still globalized at E164 in models of translation however once we hear ringback the display on the phone IP poster "private." Can someone please advise if there is an equivalent in SIP to the command no service additional h225 - prevent cid--update.

  These are the CUBE CCSIP debugging messages:

  000267: 10:38:35.506 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  GUEST sip:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  Supported: timer, resource-priority, replaces
  Min - SE: 1800
  User-Agent: Cisco - CUCM8.6
  Allow: PROMPT, OPTIONS, INFO, BYE, ACK, CANCEL, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY
  CSeq: INVITE 101
  Expires: 180
  Allow-events: presence, kpml
  Support: X-cisco-srtp-fallback,X-cisco-original-called
  Cisco-Guid: 1488217344-0000065536-0000054240-1747719340
  Session time-out: 1800
  P a claimed identity: 'CR Test MAN - 3022852' <> [email protected]/ * / >
  Privacy: id
  Remote-Party-ID: 'CR Test MAN - 3022852' <> [email protected]/ * / >; left = call; screen = yes; intimacy = uri
  Contact:
  Max-Forwards: 70
  Content-Type: application/sdp
  Content-Length: 219

  v = 0
  o = CiscoSystemsCCM-SIP 25280178 1 IN IP4 172.20.44.104
  s = call SIP
  c = IN IP4 172.20.255.249
  t = 0 0
  m = 30088 audio RTP/AVP 8 101
  a = rtpmap:8 PCMA/8000
  a = ptime:20
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15

  000268: 10:38:35.522 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  GUEST sip:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54613BD
  Remote-Party-ID: 'CR Test MAN - 3022852' <> [email protected]/ * / >; left = call; screen = yes; intimacy = full
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  Supported: 100rel, timer, resource-priority, replaces, sdp-anat
  Min - SE: 1800
  Cisco-Guid: 1488217344-0000065536-0000054240-1747719340
  User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
  Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
  CSeq: INVITE 101
  Time stamp: 1443778715
  Contact: <> [email protected]/ * /: 5060 >
  Expires: 180
  Allow-events: telephone-event
  Max-Forwards: 69
  Session time-out: 1800
  Content-Type: application/sdp
  Content-Disposition: session; handling = required
  Content-Length: 250

  v = 0
  o = CiscoSystemsSIP-GW-UserAgent 1367 2609 IN IP4 172.23.255.99
  s = call SIP
  c = IN IP4 172.23.255.99
  t = 0 0
  m = audio RTP/AVP 8 101 17778
  c = IN IP4 172.23.255.99
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = ptime:20

  000269: 10:38:35.522 Oct 2 UTC: / / 697 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP/2.0 100 trying
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Allow-events: telephone-event
  Server: Cisco-SIPGateway/IOS-15.4.3.M3
  Content-Length: 0

  000270: 10:38:35.534 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 100 trying
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54613BD
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Time stamp: 1443778715
  Content-Length: 0

  000275: 10:38:38.422 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 183 during the Session
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54613BD
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Contact: <> [email protected]/ * /: 5060 >
  Allow: INVITE, ACK, CANCEL, BYE, REGISTER, REFER, INFO, SUBSCRIBE, NOTIFY, PRACK, UPDATE, OPTIONS, MESSAGE, PUBLISH
  Need: 100rel
  RSeq: 26192
  Content-Length: 235
  Content-Disposition: session; treatment required =
  Content-Type: application/sdp

  v = 0
  o = 10406 9594 Sonus_UAC IN IP4 192.168.200.29
  s = SIP multimedia tools
  c = IN IP4 192.168.200.4
  t = 0 0
  m = 21708 audio RTP/AVP 8 101
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  a = ptime:20

  000276: Oct 2 UTC 10:38:38.426: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP PRACK:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54824BE
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  CSeq: 102 PRACK
  Grid: 26192 101 INVITE
  Allow-events: telephone-event
  Max-Forwards: 70
  Content-Length: 0

  000277: 10:38:38.430 Oct 2 UTC: / / 697 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP/2.0 183 during the Session
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
  Allow-events: telephone-event
  Remote-Party-ID: <> [email protected]/ * / >; left = called; screen = yes; intimacy = full
  Contact: <> [email protected]/ * /: 5060; transport = tcp >
  Supported: sdp-anat
  Server: Cisco-SIPGateway/IOS-15.4.3.M3
  Content-Type: application/sdp
  Content-Disposition: session; handling = required
  Content-Length: 250

  v = 0
  o = CiscoSystemsSIP-GW-UserAgent 7824 1232 IN IP4 172.23.255.99
  s = call SIP
  c = IN IP4 172.23.255.99
  t = 0 0
  m = audio RTP/AVP 8 101 17776
  c = IN IP4 172.23.255.99
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = ptime:20

  000278: Oct 2 UTC 10:38:38.442: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54824BE
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: 102 PRACK
  Content-Length: 0

  000279: Oct 2 UTC 10:38:38.922: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54613BD
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Accept: application/sdp, application/isup, application/dtmf, dtmf-relay application, multipart/mixed
  Contact: <> [email protected]/ * /: 5060 >
  Allow: INVITE, ACK, CANCEL, BYE, REGISTER, REFER, INFO, SUBSCRIBE, NOTIFY, PRACK, UPDATE, OPTIONS, MESSAGE, PUBLISH
  Require: timer
  Supported: timer
  Session time-out: 1800; recycling = uac
  Content-Length: 235
  Content-Disposition: session; treatment required =
  Content-Type: application/sdp

  v = 0
  o = 10406 9594 Sonus_UAC IN IP4 192.168.200.29
  s = SIP multimedia tools
  c = IN IP4 192.168.200.4
  t = 0 0
  m = 21708 audio RTP/AVP 8 101
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = sendrecv
  a = ptime:20

  000280: Oct 2 UTC 10:38:38.926: / / 697 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP/2.0 200 OK
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  CSeq: INVITE 101
  Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, VIEW, SUBSCRIBE, NOTIFY, INFO, REGISTER
  Allow-events: telephone-event
  Remote-Party-ID: <> [email protected]/ * / >; left = called; screen = no; intimacy = off
  Contact: <> [email protected]/ * /: 5060; transport = tcp >
  Supported: replaces
  Supported: sdp-anat
  Server: Cisco-SIPGateway/IOS-15.4.3.M3
  Session time-out: 1800; recycling = uac
  Require: timer
  Supported: timer
  Content-Type: application/sdp
  Content-Disposition: session; handling = required
  Content-Length: 250

  v = 0
  o = CiscoSystemsSIP-GW-UserAgent 7824 1232 IN IP4 172.23.255.99
  s = call SIP
  c = IN IP4 172.23.255.99
  t = 0 0
  m = audio RTP/AVP 8 101 17776
  c = IN IP4 172.23.255.99
  a = rtpmap:8 PCMA/8000
  a rtpmap:101 telephone-event/8000 =
  a = fmtp:101 0-15
  a = ptime:20

  000281: 10:38:38.926 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP ACK:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK549768
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  Max-Forwards: 70
  CSeq: 101 ACK
  Allow-events: telephone-event
  Content-Length: 0

  000282: 10:38:38.934 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  SIP ACK:[email protected]/ * /: 5060; transport = tcp SIP/2.0
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ece183b5108
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  Max-Forwards: 70
  CSeq: 101 ACK
  Allow-events: presence, kpml
  Privacy: id
  Content-Length: 0

  000283: 10:38:45.426 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  SIP BYE:[email protected]/ * /: 5060; transport = tcp SIP/2.0
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ed866258a9
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  User-Agent: Cisco - CUCM8.6
  Max-Forwards: 70
  Privacy: id
  P a claimed identity: 'CR Test MAN - 3022852' <> [email protected]/ * / >
  CSeq: 102 BYE
  Reason: Q.850; cause = 16
  Content-Length: 0

  000284: 10:38:45.430 Oct 2 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Envoy:
  SIP/2.0 200 OK
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ed866258a9
  From: "Anonymous" <> [email protected] / * />; tag=25280178~17337b8f-a50d-45a4-8f3a-f81dd5cc611b-41234378
  To: <> [email protected]/ * / >; tag = 2E9A10C-CB9
  Date: Friday, October 2, 2015 09:38:45 GMT
  Call ID: [email protected] / * /
  Server: Cisco-SIPGateway/IOS-15.4.3.M3
  CSeq: 102 BYE
  Reason: Q.850; cause = 16
  P-RTP-Stat: PS = 351, OS = 56160, PR = 342, OR = 54720, PL = 0, JI = 0, THE = 0, 6 =
  Content-Length: 0

  000285: 10:38:45.434 Oct 2 UTC: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Envoy:
  SIP BYE:[email protected]/ * /: SIP-5060/2.0
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54A13C3
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Date: Friday, October 2, 2015 09:38:35 GMT
  Call ID: [email protected] / * /
  User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
  Max-Forwards: 70
  Time stamp: 1443778725
  CSeq: 103 BYE
  Reason: Q.850; cause = 16
  P-RTP-Stat: PS = 342, OS = 54720, PR = 480 OR = 76800, PL = 0, JI = 0, THE = 0, 6 =
  Content-Length: 0

  000286: Oct 2 UTC 10:38:45.454: / / 698 / 58 B 465000000, SIP, Msg, ccsipDisplayMsg:
  Received:
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 172.23.255.99:5060; branch = z9hG4bK54A13C3
  From: "anonymous" <> [email protected]/ * / >; tag = 2E995B0-BF7
  To: <> [email protected]/ * / >; tag = gK0291ecd6
  Call ID: [email protected] / * /
  CSeq: 103 BYE
  Content-Length: 0

  Carl Ratcliffe

  Preston-Lancashire-England

  Carl,

  Understand why you are getting the private sector in State of ringtone will lead us to solve this problem much more efficiently.

  If we look at the 183 Session progress sent to CUCM...

  +++ See here this part called privay = full (meaning private)

  Envoy:
  SIP/2.0 183 during the Session
  Via: SIP/2.0/TCP 172.20.44.104:5060; branch = z9hG4bK3e5ecc3bcebb92
  -
  Remote-Party-ID: <> [email protected]/ * / >; left = called; screen = yes; intimacy = full

  Now that we know this, I would take a different approach to solve this rather than disable remote-party id in total...

  Here is my proposed solution...

  SIP-class voice profiles 3
  answer 183 sip remote-Party-ID header change "" "<>[email protected]"/ * / > ""

  Then you must apply the sip profile at the foot of basketing od the call... IE cucm cubed

  Dial-peer telephony voip xxx
  profiles of sip voice-class 3

  Now, when the appeal is as a ringtone, the full display will be + 44... What was initially called number.

• Sip Trunk design question

  Hello

  I have a requirement to pass an h323 to SIP environment environment. I'm looking for good practices, especially around security. I have 2 servers CUCM (8.5) in cities separated for redundancy. I have also 2 voice gateways which, at the present time, h323 to the PSTN, are each located in different cities.

  My requirements are:

  1. create a sip trunk instead of the supplier of the use of PRI.

  2 If the Wan link fails on a gateway provider, router replacing in the other location should be able to receive installation messages and if a user connects via extension mobility, should be able to answer the call.

  Is there a simplified design docos on for this? I hesitate to create a SIP trunk directly to the supplier for safety, thus thinking to end the call on the routers of voice with the CUBE. I am sure that it is managed from the factory and would appreciate comments.

  See you soon!

  Pieter

  Simple answer use ALWAYS the CUBE.  With IOS 15.1 T and more you have security against fraud free of charge that you can use to restrict which can address IP contacted the CUBE, that's all you need.

  HTH,

  Chris