ISDN GW 3241 - security SIP

Similar Questions

• Rules of ISDN GW 3241

  Hi all

  I have set up next week an ISDN gw 3241.  I've done a few readig on it (the user guide and other cisco docs).

  In my implementation we hane one ISDN gw with 1 E1 (several DID).  several points of video endpoints (VEP) will use this ISDN gw.

  All IP VEP are registed to the VCS.

  I want to set it up this way:

  IP for ISDN

  No matter what IP VEP, when calling IE 88XXXXXXX (88 is the prefix of video isdn gw), the call will be routed to the ISDN GW.

  ISDN to IP

  If the remote call of VEP ISDN GW number + 44335xxxx01, call get routed to IP VEP1

  If the remote call of VEP ISDN GW number + 44335xxxx02, call get routed to IP VEP2

  If the remote call of VEP ISDN GW number + 44335xxxx03, call get routed to IP VEP3

  If another number in this range is this is to say composed + 3245xxxx06, appeal to be rejected.

  Can someone share some shots of a configuration of ISDN gw with commentary and similar rules? Thanks in advance.

  Kind regards

  Hi friend

  With regard to the fraud free, I usually use a simple methodology that brings a security level pleasant without requiring a complex configuration. CPL is a nice option too, but it's too complex in some environment, and it is difficult to manage and troubleshoot in my opinion.

  Normally I restrict access to the ISDN gateway by only search rules. Try this:

  1. Let's say your code access to the PSTN is 9. If your end points would call a PSTN number dial 9 + number.
  2. Let's say that your ISDN gateway is to register the prefix 9 for VCS, so that when you dial 9XXXX the call is routed to the ISDN gateway that is registered in the local area of VCS
  3. Create a rule to search specific to ISDN calls, see example below, examine the marks:

  

  Please, follow all the instructions, I mentioned above, it is so important.

  If you have several access codes to the PSTN, you can create several rules of research as a result of this procedure, or you can use a regex to sum up the roads. For example, let's say you have 99, 98 and 97 as codes PSTN, your template string may be (99 | 98 | 97) \d*. This route matches any number including the prefix is 97 or 98 or 99.

  Another important consideration is, if you have PSTN access codes (not having not makes no sense), you need to design a numbering plan that avoidsmismatch between your access codes to the PSTN and your alias and routes. It is extremely important! For example, the suggestion that I placed above, you should not have any endpoint whose alias begins in 9 , because it would result in a conflict with your ISDN search rules.

  Prior to performing any configuration, save time for thinking and planning. I also suggest make you several test after the implementation of your ISDN integration with free fraud prevention.

  I hope that this is useful.

  Best regards

  Paulo Souza

  Please note the answers and mark it as "answered" as appropriate.

• 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

• Configure several voice prompt on ISDN GW 3241

  I would like to know if it is possible to configure the ISDN GW 3241 with several guest voices.

  I want to configure an auto attendant in English on a phone number and automatic monitoring in French on another phone number.

  Concerning

  Fred

  No, there is only a single auto attendant on an ISDN 3241.  You will need two of them if you wanted two different languages.

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

• 3241 GW ISDN with CUCM 10.5 or VCS

  Hi all

  What is the best practice to deploy ISDN GW 3241.

  It connects with CUCM or VCS control?

  How to connect it with CUCM or VCS?

  Kind regards

  There are some basic steps in the Quick Start Guide and there is more information on the different ways to configure it in this thread: https://supportforums.cisco.com/discussion/11580051/isdn-gw-3241-admin-config-guide

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

• SAI 243 x as SIP <>modem router ISDN with iLBC

  Hi all

  I have a couple of SAI 243 x (specifically IAD2431 and IAD2432) running IOS 15.1 (4) M5 that I use as SIP for ISDN PRI E1 gateways. They are configured with POTS by a dial peer for ISDN and a couple of VoIP dial-peers according to the new number called ISDN. He works very well with G.711 A-law-to-end, but now I'm trying to work with iLBC and having problems.

  My dial-peer configuration looks like this:

  voice POTS dial-peer 1

  Description peer ISDN30

  translation-profile incoming incoming bt

  service logon

  destination-model. T

  alert progress_ind-band 8

  No Strip number

  direct line to inside

  port 1/0:15

  Forward-digits all the

  !

  Dial-peer voice 3 voip

  your reminder alert-non-PI

  Description peer to number of bearer

  huntstop

  service logon

  destination-model 01234567890

  RTP payload type nte your 102

  RTP payload type comfort-noise 13

  session protocol sipv2

  session target ipv4:1.2.3.4:5060

  codec voice-class 1

  numbers-fall of DTMF-relay rtp - nte

  IP qos dscp cs3 signaling

  CLID network provided

  replacement CLID name

  !

  Dial-office of communications telephone voip 65008

  your reminder alert-non-PI

  Description peer for the main range

  huntstop

  service logon

  destination-model 01234567 [1-8].

  RTP payload type nte your 102

  RTP payload type comfort-noise 13

  session protocol sipv2

  session target ipv4:1.2.3.4:5060

  codec voice-class 1

  numbers-fall of DTMF-relay rtp - nte

  IP qos dscp cs3 signaling

  CLID network provided

  replacement CLID name

  Another relevant config:

  voice, send rtp-received

  !

  voice service pots

  keepalive RTCP

  !

  voip phone service

  list of approved IP addresses

  IPv4 1.2.3.4

  keepalive RTCP

  Standard DTMF-interworking

  Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 relief pass through g711alaw

  SIP

  udp tcp transport switch

  No anat

  block 183 present sdp

  !

  voice class codec 1

  mode of ilbc codec preference 1 30

  codec preference 2 g729r8

  g711alaw preferably 3 codec

  !

  Now, when I place a call from the ISDN through the gateway, the SDP, he offers to my SBC is:

  v = 0

  o = CiscoSystemsSIP-GW-UserAgent 7963 1810 IN IP4 6,7,8,9

  s = call SIP

  c = IN IP4 6,7,8,9

  t = 0 0

  m = audio RTP/AVP 116 18 8 101 13 18710

  c = IN IP4 6,7,8,9

  a = rtpmap:116 iLBC/8000

  a = fmtp:116 mode = 30

  a G729/8000 rtpmap:18 =

  a = fmtp:18 annex b = No.

  a = rtpmap:8 PCMA/8000

  a rtpmap:101 telephone-event/8000 =

  a = fmtp:101 0-16

  a = rtpmap:13 CN/8000

  It is very good - both ends of iLBC agreement and the call comes. However, if I make a call on the side SIP through ISDN, I offer you this:

  v = 0

  o =-78207127932020 78207127932020 IN IP4 1.2.3.4

  s = -.

  c = IN IP4 1.2.3.4

  t = 0 0

  m = 24132 audio RTP/AVP 110 8 127

  a = rtpmap:110 iLBC/8000

  a rtpmap:127 telephone-event/8000 =

  a = fmtp:110 mode = 30

  a = silenceSupp: suite.

  a = ptime:20

  Then I'm a 200 OK with that:

  v = 0

  o = CiscoSystemsSIP-GW-UserAgent 9977 8349 IN IP4 6,7,8,9

  s = call SIP

  c = IN IP4 6,7,8,9

  t = 0 0

  m = 17650 audio RTP/AVP 8 127

  c = IN IP4 6,7,8,9

  a = rtpmap:8 PCMA/8000

  a rtpmap:127 telephone-event/8000 =

  a = fmtp:127 0-16

  a = ptime:20

  This means that we use G.711 A - Law. If I've got this remove the offer and propose simply iLBC, the IAD rejects the appeal.

  Clearly, it takes care of calls using G.711 on the ISDN and iLBC side SIP - how to persuade to allow calls made from the side SIP at work and calls made from the side of ISDN?

  Thank you very much in advance for your suggestions!

  Sean

  You should have "incoming called-number." under voip RFP.

• Gateway ISDN without control of VCS

  Hi team,

  We have a Cisco CUCM v 10.5 deployed and all voices and video endpoints registered to CUCM only, now I want to integrate Cisco ISDN GW 3241 CUCM, is it possible? (as the way we add voice gateway) or ISDN GW work will remain alone (by registering on the internal Porter)?

  Please suggest...

  My recommendation would be to implement the ISDN gateway as a standard SIP trunk and configure your dial plans to ensure that the PSTN calls get routed to the ISDN gateway. You should be able to use TCP or TLS Protocol (assuming you get certificates set up correctly), but I would recommend using only "early" in the SIP profile.

• SX20 and link ISDN with TC6

  Hi guys,.

  I recently upgraded a SX20 and link ISDN to the new TC6 and after the upgrade, I can't make ISDN call.

  I can see and manage the ISDN link from the SX20 configuration web page, but the status of the ISDN connection is: error, SIP not saved.

  I use H323 network and I don't use SIP between SX20 and ISDN link and with the old version, I could receive and make calls. Now when I try to dial the ISDN using the new interface TC6 endpoint indicates that it does not find the ISDN link.

  Thank you for your help,

  Adrian

  Hi Adrien, Valentine's day.  Theres something happens with outgoing enabled on the link and the Codec for this, and we have two CDETS opened on them.  There are always SIP service is enabled on the Codec with link and link itself.  Codec uses IPV6 local links for the link itself.  Even if the sip xstatus on the codec has its inactive etc, the codec message still SIP REGISTER to the device registration link.  Link runs a FWD SIP process that handles the request.  Sorta like a mini SIP registrar if you want.  Codec will send the message to REGISTER using link-local IPV6 address.  Example of message is less than 200 Ok.

  77582.64 SipPacket PacketDump: Proto: SIP, name: REGISTER sip: [0000::000:0000:0000:cd39]: 15070 SIP/2.0, Direction: outgoing, RemoteAddress: [0000::000:0000:0000:cd39]: 15070, GroupEntity: b27d07c1f5f298c492db68040b320ca6, time: 77582634 content:!

  77582.64 SipPacket REGISTER sip: [0000::000:0000:0000:cd39]: 15070 SIP/2.0

  77582.64 SipPacket Via: SIP/2.0/TLS [fe80::250:60ff:fe83:c52a]: 5061 branch; = z9hG4bK45a7681a50ebbdc5b98e2f91bff6b438.1; rport

  77582.64 SipPacket Call-ID: b27d07c1f5f298c492db68040b320ca6

  77582.64 SipPacket CSeq: 47083 REGISTRY

  77582.64 SipPacket Contact:

  «': 5061; transport = tls >; + sip.instance =»

  77582.64 SipPacket of:; tag = 116a994ac2281c72

  77582.64 SipPacket to:

  77582.64 SipPacket Max-Forwards: 70

  Road to SipPacket 77582.64:

  77582.64 SipPacket allow: INVITE, ACK, CANCEL, BYE, update, INFO, OPTIONS, CONSULT, INFORM

  77582.64 SipPacket User-Agent: TANDBERG/518 (TC6.0.0.876266)

  77582.64 SipPacket expires: 3600

  77582.64 SipPacket authorization: Digest-nonce = "d9605a7da8be22af8c0e7f52b3de4a5f5125fcd3", domain = "MARCIE", qop is auth, username = "djkvvknn", uri is "sip: [0000::000:0000:0000:cd39]"

  ", response ="3b27b753cb9bcf38cc2431b767bce643", algorithm is MD5, nc = 00000002, cnonce ="84addca2a68f6c5f9b657a94653a4bae"

  77582.64 SipPacket supported: replaces, timer, 100rel, gruu, way, way out

  77582.64 SipPacket Content-Length: 0

  77582.64 SipPacket

  77582.64 SipPacket >!

  -----------------------------------------------------------------------------------------------

  77582.65 SipPacket SIP/2.0 200 OK

  77582.65 SipPacket Via: SIP/2.0/TLS [fe80::250:60ff:fe83:c52a]: 5061 branch; = z9hG4bK45a7681a50ebbdc5b98e2f91bff6b438.1; rport = 41246; receipts = 0000::000:0000:0000:c52a

  77582.65 SipPacket Call-ID: b27d07c1f5f298c492db68040b320ca6

  77582.65 SipPacket CSeq: 47083 REGISTRY

  77582.65 SipPacket of:; tag = 116a994ac2281c72

  77582.65 SipPacket to:; tag = e84d13204776ebc2

  77582.65 SipPacket server: fwd

  77582.65 SipPacket Contact:; (+ sip.instance =""

  77582.65 SipPacket Content-Length: 0

  77582.65 SipPacket

  77582.65 SipPacket >!

  -------------------------------------------------

  I removed my own addresses, so you get the picture here.

  It is the same for the link.  Link will send REGISTRY as well to himself.

  Even if your using H320 as outgoing Protocol call, messaging between the codec and the liaison unit is still SIP as the outgoing call is still SIP of the codec for the link even if you chose H320.

  You can see all happening all by running: Journal ctx sippacket debug 9 and connect the output to see the messaging back and forth between devices.  This is why the Service SIP on the codec cannot be disabled.

  The outgoing enabled on both devices and it records not as what Adrian has met is odd. So the question has been filed and is under investigation.

  CSCue57128 - SIP in a strange state connecting to link ISDN (Codec)

  CSCue73049 - outbound SIP on ISDN link activation causes lose her registration (link)

  These are new, so the visibility may not be there yet.

  If you see a problem with the link is not not able to register, check to see if outgoing SIP are enabled on the link.  If so, turn it off and start.

  Do the same on the codec as well.  Disable outgoing SIP and restart the computer.  There may be some cases, you do not see this, as the investigation continues with the bug above two ID's

  I hope this helps.

  VR

  Patrick

• Duplication of the SIP URIs with prefix GW

  Hello Techies,

  I recorded an ISDN gateway with VCS control with prefix 7. So, when the user wants to dial call ISDN, it includes 7followed by number and call routes to the ISDN gateway which bands of 7 and sends the call to the public network.

  However, when the same user dials SIP URI [email protected] / * /, instead of going to VCSexpressway call, he goes to the ISDN gateway.

  How can I work around this problem?

  thnx

  -wala

  Wala Hi, welcome to the Cisco support community!

  You should implement a dial in VCS who avoid that kind of shift. I don't know what rules of research and transformations that you configured, but here are some points you should consider:

  1 - alias [email protected] / * / not must be sent to the local VCS control, if this happens, you probably have a search rule 'any alias' that points don't localZone.GetDaylightChanges, that is not correct, you must create specific search models rules correspond to your alias of local endpoints registered

  2. with regard to the ISDN gateway, you should have a search rule specific to the call of the road to ISDN. For example, SIP endpoints reached ISDN, you should have a rule of research pointing to localZone.GetDaylightChanges whose model is (\1 7\d +) @localdomain.com and replace. This remove only the domain part of the URI and send only the 7XXXX recorded localZone.GetDaylightChanges ISDN bridge. In this case, I am assuming that you have a transformation with the model ([^ @] *) and replace-[email protected] / * /. If you have not this turn, you must create a different search rules to match route endpoints H323 ISDN dial only the number, it would be something like this: 7\d + and route to the local area.

  3 - Furthermore, when the creation of roads, prefer to use regex strings of preffix and suffix, regex is the best option to avoid the incompatibility of numbering plan. Rather than use the prefix 7, use regex 7\d +. For example, the prefix 7 Games [email protected] / * /, but 7\d regex + does not work.

  4 - there is an important thing to consider when registering gateway ISDN to VCS, especially when you have VCSe. You must implement a free fraud prevention mechanism in order to avoid external users to use your system as a free phone system. You can do this by using the strategy to call him; or using the rules of search + authentication + subzones. You can take a look at this thread:

  https://supportforums.Cisco.com/message/3971947#3971947

  I hope this helps.

  Paulo Souza

  Please note the answers and mark it as "answered" as appropriate.

• ISDN-&gt; disconnected IP calls

  Hello

  We have severel endpoints (C40 and 95MXP) with difficulties to follow connected ISDN/IP calls or even connect at all. Calls that crosses disconnect between 1 to 50 seconds. There is nothing no indication that calls are being disconnected with force.

  Sometimes the phone call seems to go nowhere, but the video end point really rings a few seconds.

  When the call is connected, we can hear the voice of the caller until the call is disconnected.

  I had (31) after log out message once.

  We have an ISDN GW 3241

  This rings a Bell with anyone?

  Best regards

  André

  The PRI channel could be out of sync and restarting resynch them.

• Requirements of Cisco TelePresence C for landlines and mobile calls

  Hello everyone,

  I have questions about Cisco TelePresence C: I know that you can call to landline and mobile phones as well as these.

  I would like to know more technical details about this:

  -What else should be part of the installation of the TP C series in the case of calls on fixed or mobile? This Protocol must be used (H320 ISDN)?

  Should I also buy Cisco TelePresence ISDN Link - http://www.cisco.com/c/en/us/products/collaboration-endpoints/telepresen... or can I use a gateway and SIP protocol for fixed lines?

  Thank you for the answers.

  Inorder for you make or receive calls to PSTN/Mobile then you will need some sort of bridge. Yes the ISDN link can work if you have a few points of termination.

  If you have more endpoints and activate this cabability for all then a network infrastructure device would be best, such as the Cisco TelePresence ISDN GW 3241

• Failover of telepresence conductor

  Hello

  We have a conductor & two telepresence conference bridge.

  We are in the phase of deployment & assuming telepresence are registered in the conductor & driver is connected to 10.5 CUCM via SIP Trunk.

  In this scenario, a conductor is a single point of failure. If the driver fails, CUCM will not be able to communicate with the telepresence.

  What happens if the driver goes down? Users will not be able to make Conference calls? To reach the failover can I set up 2 different SIP trunks in CUCM, one for CUCM to the driver & second for CUCM to TP & assign priorities?

  We have a gateway ISDN CTI-3241-GWISDNK9 as well. What is the best place to deploy the ISDN gateway in network? is it connected with driver or CUCM?

  For the planning of conferences, which is also the best place for the costume of telepresence management? must connect with the driver or with CUCM?

  Kind regards

  Francisco

  Hello Francisco -.

  To answer your question: what happens if the driver goes down?

  Because CUCM sees the conductor as the conference bridge, as the telepresence server is located behind the driver, no conference that use the driver would be able to begin to be.  It would be the same as if you were using TelePresence server directly with CUCM and it should fail, no conference using the TP server would be able to start.  The same would apply even if you have used the TMS.

• Cannot change the Zoneinfo file

  I had problems with El Capitan and decided to run a permissions repair script to see if it solved the problem. It came with two executable files that can not be changed. I looked on my first mac and the same files were not executable. I have a great difficulty to try to correct the incorrect files or copy the right files the first mac to it. I tried Sudo Rm, Chmod, Chown root. All attempts have been thwarted. What else can I do other that Reinstall from scratch?

  

  It's probably the new security SIP in El Capitan, which prevents make you these changes. You are right however that these files do not have x (execute) permission turned on.

  You could wait 10.11.4 and see if she arranges otherwise you could temporarily cut off SIP to enable to fix now, or as you say erase and reinstall.

• Help setting up Dial-Peer in the CME

  Hello experts,

  I tried to set up my network voice for 2 days but no luck.

  Here is my scenario:

  CUCM

  CME

  IRIS PBX

  +++ Posted inside Config.

  CME and PBX are PRI resources; I can see active layer 1 and Layer 2 multi_Frame_Established in 'display the ISDN State '.

  SIP trunk of CUCM to established CMF.

  Cannot communicate registered CUCM to analog phones in PBX Voip phones.

  Several times analog voip calls work and then some changes in the PBX, which in fact shit once again. (Those that have been made by PBX technician)

  VoIP ext: 1003

  Analog position: 270

  !! for voip analog calls

  In CUCM:

  Routing model: 9XXX--> to the trunk of the CME

  In the CME:

  The destination model: 9... --> E1 card (with auto stripping)

  !! for analog / voip calls

  In PBX access code 4 and stripped when sending guys.

  that the means to post analogue for reach 1003 user consist 41003 and 4 will be stripped.

  In the CME:

  The destination model: 1... --> session target ipv4:

  With this configuration in the 'debugging ccapi inout"see"transfer number is NULL"and

  in the "show history of last calls 1" am see reason code "1 c" - invalid number.

  After some research, this are my ideas but not tested:

  1. I have incoming dial Exchange files corresponds to the movement of e1 card:

  voice pots Dial-peer X

  incoming called-number. T

  (2) I have to configure dtmf-relay-notify sip and dtmf-relay rtp - nte to accept the numbers of telephone service.

  (3) need to create any DIDs for incoming calls to pbx?

  Please help me on this point, how could I get from analog to voip calls and vice versa, which I think is fair? If this is not the case, what would be the alternatives? where we should use the dtmf-relay command. ?

  Thanks to all in advance.

  Current config:

  type 0 1 e1 card
  network-clock-participate wic 1
  network-clock-select 1 E1 1/0/1
  primary-qsig ISDN switch type
  voip phone service
  h323 connections allow h323
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  no service additional h450.2
  no service additional h450.3
  Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none
  SIP
  Registration Server expires max 600 min 60
  no update-callerid
  !
  voice class codec 1
  preferably 1 codec g729br8
  codec preference 2 g729r8
  g723ar63 preferably 3 codec
  g711ulaw codec preference 4
  g711alaw preferably 5 codec
  !
  controller E1 1/0/1
  No.-CRC4 framing
  time intervals PRI - Group 1-31
  !
  Voice-port 1/0/1:15
  Capitalization in the carrier speech
  MGCP
  MGCP behavior considered range tgcp only
  MGCP comedia-role behavior no
  disable the behavior MGCP comedia-check-media-src
  disable the behavior of MGCP comedia-sdp-force
  !
  profile MGCP default
  !
  !
  !
  !
  Dial-peer voice 3 pots
  destination-style 9...
  port 0/1/1:15
  !
  Dial-peer voice 1 voip
  destination-model 1...
  session target ipv4:10.15.108.242
  !
  !
  !
  !
  access controller
  Shutdown
  !
  !
  phone service
  MAX conferences 8-6 win
  transfer full-consult system
  !
  !

  OK, lets delete all of your peers-Dial and start from scratch.

  For testing purposes, we will test with specific numbers.  We can get more general thereafter.  Try the following tips for incoming traffic.

  voice pots Dial-peer 10

  / / DESC # incoming Iris PBX #.

  incoming called-number.

  direct line to inside
  Forward-digits all the

  Then lets try this for your Call Manager Dial-Peer

  Dial-peer voice voip 20

  destination-model 501

  session protocol sipv2

  session target ipv4: CUCM_IP

  If please retest and propose some outputs fresh showing.

  debugging ccapi inout voice

  ccsip debug messages

  Debug ISDN q931

• CUCM 8 and Tandberg E20

  Hi all

  I tried and tried to record my E20 to the CUCM8.0 (3) as a third-party SIP (Advanced) with no luck.

  Someone at - it step by step for her.

  I have set up for the end user with username 1000 and digest the 12345 password

  Set up the new device as 3rd party SIP advanced

  Digest enabled in the device profile

  selected the user created earlier as a user digest

  used the Standard SIP profile for the SIP profile

  On the E20, I configured the SIP 1 profile with

  Proxy 1 - IP address of the CUCM8

  Authentication username - 1000

  Password - 12345

  URI - 1000 (I tried [email protected]/ * / _address_cucm)

  And I can't function. Can anyone help with this please. You don't see any problem with my setup. Thank you

  Adnan

  Adnan,

  I recently had to set up a SIP of Mitel 5224 CUCM 8.5 phone.  Here are the steps I did to make it work.

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  Procedure for the establishment of a third party SIP device with CUCM (5224 Mitel in this case):

  CUCM single installation:

  1. create a SIP profile for the third-party SIP phone that is a copy of the Standard SIP profile

  2 create a copy of the profile for the third-party SIP phone advanced security SIP Phone and check the authentication Digest

  Installation of CUCM telephone:

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  (Basic requires 3 DLU gives you 1 line. Advanced requires 6 DLU and gives you up to 8 lines with video)

  a. use the SIP and the security profile you created previously

  b. set the phone user digest to the end user of the phone

  c. create a line for the phone and note the extension

  2. find the end user you want assigned to a phone in CUCM

  a. set their Digest credentials (this can be critical assigned to be the same password for all users, but there's a security risk)

  b. associate phone end-user normally

  Third-party SIP telephone phone installation (this is specific to a 5224 Mitel therefore do everything that is needed for your phone):

  1. Phone start pressing down on the *SUPERKEY* (blue button)
  2. Use the phone keys to SIP mode and the phone will restart
  3. Find the IP address of the phone and connect on the phone admin page (sup: admin p: 5224)
  4. Select the Quick Start link on the left

  a. the user is the extension in the call manager

  (b) full name of the user, it's how you want the phone shows the number/user

  c. SIP to authenticate user and password are the user username and password SIP Digest in CUCM

  d. SIP Proxy server is the address IP of CUCM

  e. SIP Register Server is the address IP of CUCM

  f. press on apply

  From what I've read above maybe your security profile is not setup correctly?  Moreover, if the Tandberg camera does not make the distinction between the phone user name and the username for authentication then then end-user in CUCM name must also be the extension number. If it is to distinguish between them, you can use the user name of the phone as an extension (do not use the @) and the user name to authenticate as user name of the end user.

  I hope this helps.

  -Steven