Configure SIP URI dialing

Similar Questions

• Allow outgoing SIP URI dial on the TMS system Type: room

  Hello

  I ve added a VC-end point unsupported to the TMS with system Type: room.

  If I check the planning parameters on MSDS for this endpoint, I see that I can only plan entering, but not outgoing calls.

  We´re with the help of BBFRR and if I book this endpoint in a Multipoint meeting, I get an error of TMS routing, because the outgoing call missing clearances characteristic.

  Is there a way to activate the planning coming out for this type of endpoint?

  Is it possible to configure the device manually type for example in the MXP880 to get the outbound features?

  Is there maybe a flag in the comic book, were I can define the type of device manually?

  Thank you

  Tino

  System of 'room' type is intended for endpoints that TMS does not support, but which can be saved to the guardian or register SIP (and therefore be composed by any other endpoints). MSD cannot control endpoint, it cannot force the end point to dial a number to someone, but logical routing TMS allows incoming calls, because it assumes that the doorman / SIP registrar can route calls to it.

  Makes sense?

• 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.

• Numbering outside telepresense SIP URI

  Dear all,

  I'm new in cisco and telepresense Highway and I have a challenge to set up the system so that anyone from outside of the network should be able to manage our devices at the end of TV that are stored in CUCM presence. These are devices EX, SX and DX and the requirement is to compose these extension from any where on the outside with dialing SIP URI.

  We already have expressway-C and E, configured for the client and customers Jabber wants to use the highway to be able to do or to make calls from outside any other end point.

  This is achievable if so what would be the steps to get it.

  Hi Laurent,.

  To answer your questions:

  1 can I make a call to my Jabber client for devices SX using SIP URI.

  -  Yes. Method of numbering SIP URI is recommended, especially when the endpoints are connecting via the Internet.

  2. How can a person outside our network call on my devices EX and SX. Basically what he must dial to reach the device SX bearing the number 1234. for example [email protected] / * /.

  -  An endpoint outside the network can reach your end point within your network through the highway. Any device on Cisco Unified CM can be reached during
  the Internet by alphanumeric numbering assigned SIP URI or the required directory number (DN) using the <+E.164 number>@domaine. He is the solution provided by the side of the highway and track Express Core to extend your audio network seamlessly to mobile users, wherever they are.

  3-Yes, I need to configure anything specific for this thing.

  -You need set up a crossing secure area between your Core Expressway and the edge of the highway. Configuration is simple. Just follow the deployment guide in the link that I sent you. Refer to page 15 and the following procedures.

  Kind regards

  ACE

• TMS conference scheduled automatically connect via CCT and TS do not work whit SIP URI

  Hello

  For now I'll set up a new infrastructure of CMR. All components of telepresence have the lates software. CUCM 10.5.2.

  -MX and SX endpoints are recorded to the CUCM

  -CUCM a SIP trunk to driver

  -Ad.Hoc conferences and Rendezvous work

  -TS is managed by the conductor

  -MX and SX Endpoints, conductor and TS have been added to the TMS

  Now the problem. I'll put up a new conference at the TMS and the driver will be added automatically. Now I start the Conference and the conductor trying to call the endpoints of Pentecost sip uri format and the call fails. I thought that the driver uses the trunk for appointment in order to dial a number to cucm. CSS entering the CUCM site has been configured. Do I need something special on the driver or TS? In conclusion called Pentecost this format + 55228713030 fine work and call the Pentecost sip uri [email protected] / * / fail.

  They have a few clear example of aliases and auto dial configuration for this conference?

  If you have need of newspapers or something else made me know.

  Is there a way you can use a FULL domain for the domain part of the address that represents CUCM?

  Conductor is by adding the address of the place of CUCM configured under configuration Conference > locations.  I don't know the conditions, but it's because tenayo is not a field is full, or it is not routable.

• Supported SIP URI characters

  Hello

  I'm currently setting up interoperability between Cisco control VCS X7.1 and CUCM V7.1.3.

  The IPT dial plan consists of site prefixes and area codes that make up the full DN phones, this unique name is also starts with a #.

  For example #0044014761

  My problem is (I think), that # is a character in charge of SIP URI that is exposed in the deployment guide. I tried to use 35%, which is the ASCII value for #; Oddly enough, when you use that I can solve the alias against the CUCM SIP trunk, but when you attempt to dial for an endpoint, it fails.

  I know that my SIP trunk is OK, as I can carry 9.* by it on the break-out CUCM PSTN calls.

  Anyone encountered this problem before, or been able to work around it?

  Thank you, if

  Sent by Cisco Support technique iPhone App

  For your reference, it was similar (quite well) discussion there, a month

  https://supportforums.cisco.com/thread/2152374?tstart=210.

• SIP URI B2B calls via CUCM and Highway

  Hi all

  Cannot find out how to configure the SIP URI from SX codecs, recorded calls on CUCM.

  I CUCM and IM & P 10.5.1 a pair of X8.5.2 expressway with two areas of course - for B2B calls, and the external jabber.

  I want to make calls of SX-codecs outside via SIP, by using the URI field, for example, [email protected] / * / or [email protected] / * /. How can I configure that?

  I already have the routing model, which is the SIP Trunk for Exp - C, for H323 calls by ip address. Is it possible to make a routing model to call the SIP URI?

  You can see my deployment in this image.

  fghj.PNG

  

  Best wishes

  Igor

  Hi Igor

  These guides will help you resolve your issues:

  http://www.Cisco.com/c/dam/en/us/TD/docs/voice_ip_comm/Expressway/config_guide/x8-5/Cisco-Expressway-SIP-trunk-to-unified-cm-deployment-guide-CUCM-8-9-10-and-x8-5.PDF

  https://supportforums.Cisco.com/ru/video/12362936

  BR Oleksandr

• Change caller id / name of the SIP URI of VCS calls

  Hello

  I have clusted CUCM 9.1 with a SIP to a VCS - C trunk.  When making calls, the calls show a caller id of [email protected] / * / address, for example [email protected] / * /.  I have a configuration URI directory on the phone that generates the call, but now it shows the URI.  Incoming calls from SIP URI properly.

  Thank you

  Hello

  On the trunk SIP for VCS, under outgoing calls, there should be an adjustment: vocation and connected part format, change it to URI deliver only partially connected. Also, under the profile SIP assigned to the SIP trunk to VCS, there should be a setting use entirely field in queries, check this box. The SIP profile should also be attributed to the device SIP calls.

  HTH, if yes, please rate.

• After adding a subscriber to the CUCM, Sx20 Sip URI field empty cluster

  Hello

  After adding a subscriber to CUCM Cluster, all the restarted cluster devices, then sip uri field sx20 features seemed to "Null", simply enter the SIP registrar one by one for all devices sx20 IP.  Anyone else facing this problem? In my view, is not normal behavior.  What are the causes of this problem? Waiting for your opinions.

  CUCM system version: 9.1.2.13057 - 1

  The SX20 software version: TC7.2.0.aea6651

  //

  Thanks in advance

  Hi Dani

  Greetings of the day!

  Ideally this should not take place. I suspect this is something at the end of Manager at the origin of the problem.

  You make changes to the device pool after adding Subscriber to CUCM Cluster?

  Modifications were made to the Group Manager call?

  Replication between the call manager Publisher and Subscriber is any good?

  The SX20 is supplied by TMS devices?

  You should check the logs of provisioning of the endpoint to enter the cause for it.

  Concerning

  Radhika Sajan

• URI dialing problems

  I started a new job and have been invited to support our video-conference equipment, I have limited expereince with kit v/c, however I have several years expereince with network & unix, so please take it easy.

  I have a C20 of Tandberg & C60, which are both on the public ip, I don't have a VCS or a bridge of some sort. I can dial directly to any intellectual property, and they can call me directly. However, I can not dial all numbers that require a string of URI style ([email protected] / * /).  Just quick call fails and no reason given.

  I have a MSD Setup who can see both units, I can also connect to the web interface, the tsh & the command-line as root on both units.  I googled about as much as I can, but cannot understand what may be the problem. If anyone can point me in the right direction as to what file newspaper or the config setting I should be looking, that would be greatly appreciated.

  Stephen Bancroft wrote:
  I have been testing from the C60 which has software version TC4.1.0.247017
  

  And there is your problem...

  You cannot use the URI dialing with software versions prior to TC4.2 unless the system is registered with a Gatekeeper.

  Cisco only introduced full support for H.323 Annex O in TC4.2 - you'll need to upgrade your system.

  As is the case, you can dial IP address.

  If you do not have a service contract in place, you will still be able to upgrade the device as per the below security advises:

  http://Tools.Cisco.com/Security/Center/content/CiscoSecurityAdvisory/Cisco-SA-20130619-TPC

  Contact TAC for the release key etc.

  /Jens

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

• Cisco vcs 'Phantom' or false attempts at URI dialing

  A customer reported that there are many attempts to appeal to non-registered or 'false' URIs in their control of VCS and VCS Expressway, the consecuence is the saturation of internet bandwidth customer due to the many requests.

  This has been discussed before in these forums.  Take a look at some of the following threads:

  • nuisance-h323-call-sx20
  • Ghost-calls-tandberg e20
  • sourceh323idcisco-incomingcalls
  • VCS-sip-abuse-reported

  Wayne

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

• Change the Source URI for outbound calls using VCS Expressway X8.2

  Hi all

  I would like a little help on a CPL script, that I'm working on. I lived all the examples of community support and I am always sick.

  We run on VCS X8.2.  What I've read, VCS X8.1 has introduced a few new CPL capabilities, including the node location CPL supports rewriting the regex-based source alias.  I would like to take advantage of this new feature. I'm looking for a tutorial or a direction more advanced examples using this new capability.

  Here's my question:

  I have an internal domain "CompanyA.local" and I have a public domain "Entreprisea.com". «My Cisco video endpoints register control VCS with a SIP URI of "[email protected] / * / .local'.»

  Receive calls from outside help "[email protected] / * / com" and the rules of research, I route calls properly internally.

  When I dial an external company, my endpoint is showing a video SIP URI of "[email protected] / * / .local"

  Of course, this isn't a SIP URI dial-able and should be changed to "[email protected] / * / .com'." "

  I know that this can be achieved using CPL, but I have not used this feature before.
  Any help would be appreciated.

  Please suggest an example script CPL I can review.

  I would use a Regex to cover all of my similar to the following endpoints:

  source-url-to-message-regex = "(.*) ' @CompanyA.local '"
  source-url-to-message-replace = "------[email protected] / * /"

  Thanks in advance for any help.

  oooooo, that answer above gave me the answer I was looking for. It does not work! It seems that the syntax for the xml file has changed from pre x 8 and x 8. Looks like I've been using the incorrect syntax. I was missing the AVA: in the clause to the proxy, which is my version of rule switch not working. Still don't know why the address switch did not work, but I prefer the rule-switch anyway, so no big loss for me.

  I've tested, and it is fully functional on my vm VCS x8.2 running lab

  
  
  "xmlns:TAA ="http://www.tandberg.net/cpl-extensions"
  "" xmlns: xsi = "http://www.w3.org/2001/XMLSchema-instance"
  xsi: schemaLocation = "urn: ietf:params:xml:ns:cpl cpl.xsd" >
         
           
             
               "[email protected] / * / ' >
                 
               
             
           
         
  

• CTS 1300 and Polycom

  Hello

  I installed VCS and 1300 CTS as an endpoint. When external users of polycom sip uri dial 1300 CTS, it fails.

  If someone came in this situation?

  Mike Shah

  Hi Sandy,

  I don't think you can register CTS 1300 to VCS.

  Could you please give me some clarification about infrastructure and call flow?

  Also for external SIP URI of numbering, have you configured correct of DNS SRV records for the domain.

  Kind regards

  Sagar

• Join the public network through the firewall option MCU Conference

  Hello world! Lately, we have a goal to jabber (without vpn) of internet users to join conferences within our network. Our telepresence infrastructure includes VCS, MCUS and endpoints (MX, SX). I discovered that MCU (4505) has a feature called video firewall option (under license). My question is whether it will be sufficient to achieve our goal? I've also heard that VCS expressway is used. What is the difference between these two?

  Firewall MCU option just the second NETWORK card, so you can have one with a public IP address and the second with a private IP address (essentially NAT'ed), and the internal and external clients will be connecting to a virtual meeting room common pre-configured on the MCU.

  The VCS-E (Expressway) works in tandem with your VCS - C to offer secure firewall courses. VCS - C is located on the inside of you firewall and the VCS-E is normally in the demilitarized zone.

  This allows, among other things, external systems to call your systems using H.323 and SIP URI dial, this includes clients of JabberVideo, but not Jabber client CPU.

  For Jabber clients you would deploy normally expressway-e series, see http://www.cisco.com/c/en/us/products/collateral/unified-communications/expressway-series/datasheet-c78-730478.html and http://ciscocollab.wordpress.com/2014/01/29/deploying-collaboration-edge/ - whether it will work with your existing deployment is another story, you might be better off ultimately to something similar to Pexip or Acano - both were developed by former TANDBERG/Codian/Cisco engineers.

  /Jens

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

• Integration of TANDBERG 1700 MXP configuration with CUCM (h.323/SIP)

  I tried to configure a CUCM with TANDBERG 1700 MXP 8.5 and I could not, if someone has already done and help me with this number SIP or H323.

  Thanks in advance

  Hello

  Mxp units can be saved on cucm as endpoints (base) of third parties.

  Please find the steps to enter mxp.

  The sip on mxp as uri and sip proxy address settings.

  1. sip mode: on

  2 SIP Uri: [email protected] / * / of cucm for example: [email protected]/ * /.

  3 server discovery: manual

  4. server address: ip address of cucm.

  5 server Type: cisco

  6. transport: whether auto snack but who do not work, change to udp.

  7 TLs: off

  8: authentication sip username: user you created on cucm eg 9011 or amlesh

  9 password: use the same password configured when creating user 9011 or amlesh

  CUCM config:

  1. Select the basic profile of third party.

  2. Add the mac address of endpoint mxp.

  3. Select the appropriate sip, phone etc models profile.

  4. under user digest and owner user id select the same user, for example: 9011 or amlesh

  5. save the configuration and assign DN even that we used in the uri on mxp sip settings.

  6. also associate device mxp the user using the association on cucm devices pane.

  Please let me know if that work for you. I tried this in my lab and works.

  Thank you

  Amlesh