Gateway ISDN troubleshooting

Similar Questions

• GATEWAY ISDN and PRI

  Hello

  I have PR who include 30 phone lines, and I use with fax and telephone with 10 thereof.

  can I use free lines remain with the ISDN gateway... or do I new pri line?

  I'm gld on the will of a contribution to this number

  Thank you

  The question is how you plan to distribute calls.

  If you have a device such as a pbx that can do that is not tampering with the signal / call it should work.

  Also, make sure that the line itself supports videoconferencing.

  The easiest would be to just get a new line.

  Please remember useful frequency responses and identify useful or correct answers.

• 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.

• Questions about VCS, MCU, ISDN gateway license

  Dear Expert.

  I question of license on my next device.

  (1) I have:

  VCSC - 10 calls Non-traversal, crossing 100 calls license license

  vcse - 5 traversal calls

  If I register the mcu and the endpoint to the VCSC as the guardian h323.

  I should have a 15 (5 calls traveral and 10 calls non-traversal) sites h323 conference call simultaneously on the MCU?

  > If I record my MCUS and endpoints to the ISDN gateway as the guardian h323.

  > I should be able to do an 18 sites (1 h323 calls and ISDN 17 * 128 Kbps) conference at the same time call

  > on the MCU?

  GK embarked on telepresence ISDN Gateway has no limitation for the number of concurrent calls.

  Limitation is only for the registration number.

  As gateway ISDN support audio connection gateway exclusively, with simultaneous simple T1 (T1 = 24 b, PRI = 23 b = D), maximum number of calls is superior to '18' as audio call consume only 1 b-channel.

• 3241 GW ISDN to CUCM calls must be block

  Hi Experts

  I need your suggestion below

  One of our customer with cisco Telepresence infrastructure setup (VCSC & E, conductor, TIP Server & TMS, CUCM). CUCM and integrated VCS, we recently installed ISDN the Gulf war (3241)

  Now the client wants to block incoming and outgoing calls of CUCM to ISDN GW extension of GW ISDN extension CUCM

  CUCM numbering plan is 62XXXX series and of VCS E.164 dial Plan is 64XXXX. All products in the most recent version S/W

  someone suggest how fulfill us this requirement

  Concerning

  SR

  Hello, welcome to the Cisco support community!  =)

  There are two methods to meet your need, first using the rules of research named source areas and the second is the script of CPL. I'll give you an example on the use of the rules of research, which is the easiest way to meet your need:

  Blocking of the ISDN to CUCM

  1) register your gateway ISDN to a subarea separated in VCS control, for example 'ISDN-sub-zone '. Only the entry door must to enter this sub-area.

  2) create a search rule that fits your plan number CUCM, in your case, 62XXXX. This rule have priority over any other rules of game. Set the 'source' as the 'ISDN-sub-zone ". Use the parameter "replace" and transform the number to an unknown number. Set the 'match' 'Stop '. If your ISDN gateway will not be able to call any CUCM endpoints.

  Blocking of CUCM to ISDN

  1) the easiest is to make blocking calls using CUCM himself. You can use the CSS/Partition of CUCM features in order to refuse some end points to call ISDN numbers

  2) , but you can also use the VCS to block calls. You can use the same logic above. Create a search rule that fits your plan ISDN number, in my case, 0\d. * This rule have priority over any other rules of game. Set the 'source' as being the nearby area of CUCM. Use the parameter "replace" and transform the number to an unknown number. Set the 'match' 'Stop '. End points so that all your CUCM will not be able to call all ISDN numbers. Be aware that it will be applied to all the endpoints registered to CUCM, if you want to block only the end points, use the Partition/CSS of CUCM functions.

  Free fraud prevention

  When you implement an ISDN gateway registered in VCS, especially when you have a highway of VCS, you should consider the need to implement a free fraud prevention mechanism which will prevent external users to use your system as a phone system without. For example, if you do not implement such a mechanism, external Internet allows your VCS Expressway users to achieve your ISDN gateway and then free ISDN calls using your gateway. In addition, there is another method of fraud called "reunification", where the user external dials to your ISDN via ISDN gateway and connect to the auto attendant to recognition of the bridge, if it is enabled, then the user redial another ISDN number and then the call is routed to ISDN using your entry door, in other words, a malicious user can make a local call to your gateway and then use your gateway to recompose and make an international call, for example.

  Therefore, it is extremely important to look at a diagram of toll-free fraud prevention when implementing any ISDN gateway registered in VCS, especially when you have VCSE.

  Fortunately, Cisco provided a configuration example explaining how to block two methods of fraud without charge. Take a look at this guide from page 40:

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/VCs/config_guide/Cisco_VCS_Basic_Configuration_Control_with_Expressway_Deployment_Guide_X7-2.PDF

  I suggest you consider all of the above examples, save time to plan and then implement your restriction and ISDN toll-free fraud prevention mechanism.

  I hope this helps.

  Paulo Souza

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

• TMS smart Planner 14.2 ISDN Dialin Participants

  Hello

  With the old Scheduler, it was possible to choose ISDN for participants of callin.

  With SmartScheduler, you can choose "Forums Participants" and "audio Calling" but you can not choose ISDN so prevered Protocol is SIP.

  We have participating callin calling in via ISDN.

  Using the Scheduler by TMS you can choose 'Extern' and 'ISDN' for Nicole, you can't do with Smart Scheduler.

  With SmartScheduler the informationmail contains no ISDN Callin number.

  With the planner of TMS, the mail contains Callin number ISDN.

  But your users must use SmartScheduler because it's easier to use than the Scheduler in TMS Administration side.

  The ideals?

  Kind regards

  Martin

  Hi Martin,

  Planner smart planning only external SIP clients.  There is no ISDN remote access or outbound external.

  Potentially, you could book external ports for remote access.  If your ISDN gateway can connect to your MCU it would work.

  Gateway ISDN ISDN user calls, and then between address Conference call to MCU.

  Greetings

  Thomas

• ISDN call routing

  Hi all

  What is the best way to configure the call routing suitable for ISDN under the following conditions:

  -two gateways ISDN deployed, each at a different location,

  -the two gateways to join prefixes with the same VCS,

  -each gateway is in its own sub-area

  -each group of endpoints assigned to subareas according to their location.

  The goal is to have a defined range of endpoints in a specific location (subzone), calls are routed through the correct gateway.

  Should this be done by the removal of specific links, by policy or any other method call?

  And what about redundancy in such a deployment (if one of the bridge falling down)?

  Kind regards

  Maciek

  First of all separate the user experience with what is done on a technical level.

  (Registering a GW ISDN with a different prefix or no prefix at all) does not mean that

  the user cannot reach it or that the user can dial a specific prefix.

  Technology information is not only one way to go, there are often several ways which lead to the same goal.

  More your deployment depends on the capabilities of your team, the requirements...

  With X7.2 you can define a search based on a box named (void) rule and match your ISDN line prefix for ISDN gw1 and put other users in another area, rule of similar research with the same prefix that the user made up but this time

  matched with the other isdngw.

  For the ISDN gw you could either map via various registration prefixes or of what I prefer, put them

  each in their own areas.

  If you have these two rules search and add two more with a lower priority, it could

  work for a failover as well, but I would use a CPL or NPS server for such a scenario which

  you would better card on the case of the error.

  The VCS is ideal because it is quite simple and logical, but still has CPL policies to extend a lot.

  In any case it is not the best out of the box call control, so if you need that, look in CUCM.

• 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."

• Computer scientist - WKF

  Hello-

  We have a TelePtresence SX20 and link Cisco ISDN box.  Does anyone know if the ISDN link area is able to support two telepresence systems video via the video port using some type of switch or splitter?  That a video of telepresence system would serve both.  Thank you in advance.

  Kind regards

  WKF

  No, the Cisco ISDNLink is designed only to be associated with a single end point.

  You could do what you want to do with it, but she would need of disconnection and rematching of endpoint to another whenever you want to switch between endpoints manually - this is not necessarily a simple thing to do.

  Something like the gateway ISDN 3241 is intended to do as you want it to do, allowing to several endpoints can use an ISDN service.

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

• Receiving calls the LAN IP. Help, please

  Hello

  First time poster, but I couldn't find a similar thread.

  My C60 telepresence codec is receiving calls repeatedly every 30 seconds to a minute on my local network.  The call alert incoming says it comes from an IP address included in our premises: 10.10.10.XXX, and for each incoming call, the last 3 digits of the address IP is different (10.10.10.15, then 10.10.10.25, then 10.10.10.700, etc.).  If I leave it alone for a week, I'll be back to more than 1,000 missed calls and the machine is frozen.  Often it will be frozen every time I wake up, and I have to restart.  I have not changed the configuration at all since the latter was working fine.

  I contacted my gateway ISDN service provider first disconnect and their services to solve the problems, but the calls were still coming.  I just unplugged the Cat6 cable to see if this is the case of in the codec itself.  The calls stopped coming in and the area is sensitive again.

  Something in my local network called the box via the IP address.  Any of you have heard or seen what is happening before?  Any advice?

  Thanks in advance,

  -Rory

  Hi Rory,

  Certainly looks like a SIPVicious (or similar) probe.  These have been mentioned several times in the forums before.

  One thing that you can try on your endpoint is to put "xConfiguration SIP ListenPort: Off '.

  See Bug CSCue55239 - you can upgrade the firmware on your device too.

  PS - You may also be interested by this safety notice if you do not have a contract of service - it will help you to spend at least at TC6.1 if you are on an older version of software on your C60.

  Wayne

  --

  Do not forget to rate resposnes and mark your question as answered as appropriate.

• The numbers E.164 + and IOS voice translation rules

  Hello

  I have a cluster to CUCM 9.x with a plan of + E.164 numbering for the receivers, route patterns etc.

  I have a gateway ISDN (2911 SRI with IOS 15.3 (3) M1) which connects to the PSTN.

  The ISDN provider cannot accept the destination for the format + E.164 numbers so I need to manipulate the figures called before sending them to the PSTN switch.

  I can of course do this with CUCM party called models of translation, but I wonder if it is possible to do the translation on the bridge instead?

  The rule I tried to use below (Country Code of the United Kingdom).

  rule 1 / ^ + 44\(.*\) / / 0\1 /.

  I want it to do is + 44 at the beginning of the number of bands and replace it with 0

  for example + 441234444666 becomes 01234444666

  The rule I've tried fails because the + symbol is identified as a wildcard rather than in the number string I want to match.

  Is there a way to get around this?

  Thank you

  Hello James

  This is a regular expression, have you tried setting point (.) first then + followed by your number

  I'll try in the laboratory

  BR,
  Nadeem

  Please note all the useful post.

• 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.

• Consolidation of bridge 3241

  Can we "grouped" cisco 3241 Gateway?

  If so, is there a procedure?

  THX

  HI Firdausch.  N ° in the meaning of the word 'clustering' as MCU, cannot you cluster of ISDN GW like that.  If guardian like VCS, you can have multiple GW and use indications of availability of resources to have the GK choose what GW to send the call, etc.

  Help from H323 page:

  Send available resources:

  Select this option if you want as the gateway ISDN to inform the guard on its availability or unavailability.  This information will be used by the gatekeeper when he selects where to place calls.

  Also enter the threshold above which the messages will be sent to the doorman. The threshold is the percentage of the B channels available in use. 'Available' port has at least 1 layer upwards and the channel is not used as a D-channel/reserved channel. For example, if the threshold is 80%, the ISDN gateway informs the warden he is busy when 80 percent of its B-channels are used.

  You can use this feature in two scenarios:

  • Where multiple ISDN gateways are registered with the same prefix to dial on the doorman even plan.  When indications of the availability of resources (RAI) are configured, the ISDN gateway informs the guard when he is not available. Gatekeepers that support this feature will support ISDN gateways in the report available at the time to choose where to place the new calls
  • Where there is only one ISDN gateway, and you will want to limit the use of the gateway in IP calls. This ensures that there is always some capabilities for calls from ISDN

  When selected, the ISDN gateway informs the guard when it is unavailable (when all ports are already in use).

  The ability of the gateway ISDN to send messages of availability of resources is useful in a network where there are multiple ISDN gateways or when there are several gateway ISDN blades in a MSE.

  In an environment with multiple ISDN registered with the same goalkeeper gateways, this guard should encourage devices as available when choosing where to place the new calls.

  For example, when an ISDN gateway sends the doorman, a message indicating that it is not available, the warden tries to use a different ISDN gateway to new calls.

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

  If you use VCS I highly suggest watching

  http://www.Cisco.com/en/us/docs/Telepresence/infrastructure/VCs/config_guide/Cisco_VCS_Basic_Configuration_Cisco_VCS_Control_with_Cisco_VCS_Expressway_Deployment_Guide_X7-1.PDF

  to avoid hairpining or unauthorized use ISDN GW resources via callers calling through the solution of control/Highway etc.

  I hope this helps.

  VR

  Patrick

• 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.

• ISDN bind-l3-Manager of ccm for the PRI is not taken in charge - MGCP Gateway

  Dear all,

  Ive tried to get my (Voice Gateway) VGW recorded with my CUCM_7.1

  I must add ISDN bind-l3-Manager of ccm in my serial interface. BUT THIS COMMAND IS NOT SUPPORTED

  The only similar to this command is bind-l3 ISDN q931.

  I tried to change IOS to c3825-spservicesk9 - mz.124 - 15.T10 to c3825-spservicesk9 - mz.124 - 24.T and it gives me nothing.

  Ive checked the document - https://supportforums.cisco.com/docs/DOC-2682 and he says that a CSCed15644 bug is upward. Ive tried to do the troubleshooting steps and still I find myself in vain.

  Forward all your suggestions of urgency.

  Kind regards

  Ophelia

  'service mgcp' command to bind l3 cucm otherwise without the 'mgcp service' on the controller then configures as h323 and so linking back to cucm l3 is not allowed since the iOS think you configure the controller as a h323

  Sent by Cisco Support technique iPad App