Numbering outside telepresense SIP URI

Similar Questions

• Configure SIP URI dialing

  Hello

  I tried to activate the SIP URI dial, unfortunately with no success yet. Here you can see what I've done, maybe one of you can tell me what is missing:

  • Set up the LDAP synchronization, assign msRTCSIP-PrimaryUserAddress URI directory (because of the interoperability of Lync)
  • DNs associated with devices of CSF and BOT
  • Associate the URIs directory with directory numbers
  • The "phones" partition of directory URI alias value
  • Modify a SIP profile (named Jabber) later
    • Define interpreting string dial with 0-9, *, # and +.
    • Use the FULL domain name in the SIP application
    • Allow the presentation of bfcp
  • Assign the profile Jabber SIP devices CSF and BOT
  • Set the DNs route partition as an alias of the directory URI (phones)
  • Jabber - config.xml and change download
    • msrtcsip-primaryuseraddress in section of the book
    • true in section strategies
  • Restart the tftp server and re - open a session with Jabber

  We have 9.1.2.11900 - 12 Call Manager Server Instant Messaging and presence 9.1.1.10000 - 8 and 9.6/9.7 Jabber clients.

  Is there anything I missed or misconfigured?

  Thank you

  LACI

  Hi Labelle,

  If you put DirectoryURI in the uri section directory publication is no longer available in the client.

  Of the documentention, I got that it belongs to the policies. But if I put it here my click to call URI is always the address e-mail and not msRTCSIP-PrimaryUserAddress. So I'll do some research here.

  One thing I found is the resolution of contact section. If I signed to jabber my name did not appear. Instead, it was my msRTCSIP-PrimaryUserAddress.

  To resolve this problem, you will need to add this to jabber - config.Xml.

  true
    msRTCSIP-PrimaryUserAddress
    SIP:

  Kind regards

  Paul

• 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

• 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.

• 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

• 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.

• 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.

• 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.

• 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

• Numbering of the SIP Client IP addresses to public IP addresses

  Hello

  We are developing a new video network with VCS control and 2 highways of VCS. 1 expressway stressed the public internet address face and 2nd Expressway face partner network.

  Requirement: Internal SIP endpoint need dial IP address public (via Highway 1) and the IP address of the partner (via Highway 2). VCS control configured mode "calls to unknown IP address" to "Indirect".

  The two highways set up for 'Request to the unknown IP' mode to 'lead '.

  There is research on the VCS control rules to find the IP address.

  Priority 1 for IP address via 1st Highway

  Priority 2 for Ip address of partner via Highway 2.

  The two numbering IP addresses work, but component partner / IP address takes about 38 Sec to finish for the configuration of the call.

  When I look at the history of calls on the VCS control, I can see even for the search for the IP address of partners, 1 internet search Expressway and could not reach the IP address and then start to search on the highway of VCS partner and find the Ip address, like that it takes more time to complete the call.

  Sometimes, it is time and couldn't make a call of IP address to the IP address of the partner.

  is he kind, we can minismise the time for setting up call for the IP address of the partner numbering.

  Kind regards

  Chris

  Chris,

  try to disable SIP UDP on both motorways as this should reduce the appeal of 30 seconds preparation time.

  Hope this helps,

  Andreas

• 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?

• Problem with SPA232D + SPA302D call tones?

  I recently installed a new SPA232D ATA and SPA302D DECT handset on our network of offices, which already uses a loyal Office SPA525G2 phone. Both the SPA525 and the SPA232 register with a SIP provider hosted on separate lines. I have not all other phones (analgoue or other) connected to the SPA232. After the configuration of the unit and entering a bunch of locales, most of the things with the new phone work as expected.

  However, when I use the SPA302 (via the SPA232) handset to call the SPA525, I hear a strange tone that sounds that once followed the silence until the call is answered in charge or go to the virtual machine. The destination phone continues to ring and can be chosen as normal for the call. In addition, the tone is different of tones is entered in the regional settings of each device. Calling other phones and other results of numbers (outside my SIP account) in tones which are correct for our region. I did a very quick syslog capture with wireshark and I see a 100-> 183-> 200 SIP sequence on these calls, while other calls seem to show a 100-> 180-> sequence 200. I saw no 182 s. Not sure if this is important.

  Any idea what would cause this?

  In addition, a couple of other problems minors reagrding the SPA232D + SPA302D combo:

  -The SPA232 will not automatically save to the SIP service after a reset or power and does not attempt to reconnect after the initial failure. He recorded immediately, however, if the voice component is restarted after a config change in the GUI http. I value 0 network delay, but noticed no difference.

  -The SPA232 (or SPA302) seems to ignore most of the locale. Past to your frrequencies and cadences in config make no difference to the tones heard through the SPA302. Must these be resynced with the handset somehow?

  Any thoughts greatly appreciated.

  With our thanks,
  Nick

  the ringback tones entered into the regional settings of either device. Calling other phones and other numbers (outside my SIP account) result in ringback tones that are correct for our region. I did a very quick syslog capture with wireshark and I see a 100->183->200 SIP sequence on these calls, whereas other calls seem to show a 100->180->200 sequence. I didn't see any 182s. Not sure if that is important. 

  It IS important. Code 180 is command "generate tone locally". The callers phone will create the tone according to its (regional) configuration. Code 183 is ' I'll you sent the audio tone '-local phone generate no dial tone, just play what it comes via the audio channel called Exchange device or target her.

  You will need to check the configuration of exchange that connect call from SPA232D to SPA525 (for example call with broken tone).

• GW ISDN Cisco with VCS control

  Hi Experts,

  I'm new with Cisco ISDN Gateway 3200, and I will integrate to the existing video network managed by VCS control with the endpoints and MCUS registered as SIP.

  Here is an example of the existing video network numbering plan:

  SIP URI endpoint: [email protected] / * /

  MCU SIP URI: [email protected] / * /

  ISDN GW H323 alias: 54xxxxx

  I came up with these questions:

  1. for us intergate ISDN GW on the video network, we must save as H323 on the right VCS?

  2. to call scenario as a point endpoint SIP or MCU calling to one ISDN endpoint via video, how the call flow? Make the registered endpoint/MCU SIP point as SIP on the VCS can dial directly on this endpoint ISDN? Or they will call first the H323 number of ISDN GW recorded on the VCS then routed to an auto attendant of the ISDN GW?

  3. How about ISDN endpoint SIP endpoint?

  Please send me a sample of guide and the configuration of the ISDN GW and VCS about how we could improve the flow of calls work on the call scenarios mentioned.

  SIP---> ISDN

  ISDN---> SIP

  Thank you very much for the help.

  Best regards

  Acevirgil

  Hello

  1. For us to intergate the ISDN GW on the video network, we need to register it as H323 on the VCS right?
  

  Yes, it's true. This isn't the only method, but it is the most used and most easy way. I suggest you use it.

  2. For call scenario like from a SIP endpoint or MCU calling an ISDN endpoint via video, how's the call flow? Do the SIP endpoint/MCU registered as SIP on the VCS can dial directly on that ISDN endpoint? Or they will dial first the H323 number of the ISDN GW registered on the VCS then routed to an auto attendant of the ISDN GW?
  

  In this case, you will need to ensure interoperability the call in VCS. No matter what SIP endpoint can dial numbers ISDN, and then VCS will route the call to the gateway h323 format, only the number with any ' @domain.com ' and the call will be interoperability. The flow would be something like this:

  Point endpoint [sip] SIP----> - VCS [H323]---> Gateway ISDN

  3. How about from ISDN endpoint to SIP endpoint?
  

  The same concept is applied. ISDN gateway sends the call in H323 for VCS and interwoks VCS the call and sending SIP endpoint. Something like this:

  SIP endpoint<-------[sip]--------- vcs=""><--------[H323]---------- gateway="">

  With regard to incoming calls from gateway ISDN to VCS, there is one important thing to consider. You basically have to methods:

  • You can configure the ISDN gateway to route calls to the auto attendant, then users will be able to recompose the verse numbers of internal endpoint for the auto attendant
  • You can configure the ISDN gateway to use DID (Direct inward dialing), in this case, you create routes to the gateway which maps each ISDN number to an internal endpoint registered to the SCV

  Both methods work fine, however, when using auto attendant, it is very important to implement a scheme of toll-free fraud prevention. Take a look at this thread:

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

  Regarding the guide, it is not a simple step by step explaining, but the Administrator's guide provides a good explanation on how to configure the dial plan in the gateway and how to enter the doors of VCS. Check out this guide:

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/isdn_gw/admin_guide/isdn_gateway_printable_help_2-2.PDF

  I hope this helps.

  Concerning

  Paulo Souza

  My answer was helpful? Please note the useful answers and do not forget to mark questions resolved as "responded."

• 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] / * / ' >
                 
               
             
           
         
  

• 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.