ISDN call routing

Similar Questions

• Audio call routed to the speaker when it is docked

  Hello

  When I anchor my Razr HD (JB 4.1.2) to my real car Motorola dock, my music will be automatically forwarded to speakers of the car, which is what I want, but for incoming calls and outgoing audio passes the external speaker to phones by default. I can touch the speaker button that forwards then the audio to the speakers of the car, but how can I do audio call road to the default car speakers and not the external speaker phones?

  If it is not a setting I can change then maybe an app that will do it for me?

  It seems a little strange that he must use the phones pregnant as I thought that the idea of a dock is to have fewer entry during the conduct of the user, I'm sure that ICS was not like that.

  Thank you

  Hi xrcist,

  Another phone never worked with your car? Were you able to route calls to your front car speakers?

  See you soon,.

  Nicole

• VCS - CPL to block call routing to and from the same area of neighbor

  Hi all

  Is there a CPL rule that can be written to block calls from a neighbouring area to the same area of neighbor?

  We have problems with CUCM sent appeals to the our VCS VCS then routes the call return CUCM, causing thus "too many jumps."

  In the example

  MOVI user control VCS try to compose endpoint registered RTC CUCM, but incorrectly dials

  Appeal does not have to search for the terms of local area

  Call does not match search for VCS Expressway rules

  Call corresponds to captures all of CUCM (. *) @domain.com

  CUCM does not form DN or URI for the wrong phone number and the call is routed to the VCS control due to the SIP routing model of *. * on CUCM

  We have to capture all CUCM at the end, since we offer PSTN, as well as our users and phones have the alphanumeric URIs assigned to the DNs for Jabber, etc.

  I fear that he is not really an easy way to do it, I know because the call charged CPL first, CPL don't know what area it was intended for.

  It's probably more easily accomomplished with a slight modification to your rules of research rather than spraying anthing "@domaine" everywhere - something like:

  Anything incoming directly at @domaine in the nearby area of the local area (as is inside your organization, such as if the CUCM not found and sent it to the VCS he won't be on the CUCM, so must be local (or nonexistent) - and stop.

  And the opposite of the above - route whatever it is for @domaine of the localZone.GetDaylightChanges (because it was not found locally) to stop and the CUCM neighbour-

  And do similar for what is coming from other areas outside to route to wheverver that will be endpoints.  In this way, your call should not be bouncing back between the CUCM and VCS and create the loop that you are currently experiencing.

  Of course, if your environmetn is a little more complex than the simple CUCM and VCSes, this can be a bit oversimiplified, but could lead you in the right direction.

  Another thing to consider too - if you put in your endpoints in a directory, users can use - that will help you to prevent wrong type stuff :)

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

• 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."

• Routing of calls

  Is it possible to route 911 calls on a specific gateway with a telephone service and carry regular calls on a different gateway? I am running version CUCM system: 9.0.1.10000 - 37.

  Thank you

  Alex

  two copies

  https://supportforums.Cisco.com/discussion/12140231/call-routing

• 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.

• Selection of the correct ISDN gateway

  Hi all, I'm new to the vcs, so I'm sorry if the question could be mundane.

  We have a Cluster consisting of 2 vcs - c placed in 2 regional sites differet and one ISDN gateway for each site.

  When we try to set an ip address to isdn call it begins from the bridge of the side where there is the master of vcs, so the question is: How can make us the call out through the right site gw?

  It's the solution that we thought:

  the dial plan is 12xxx for one site and 13xxx to other sites, so a CPL could filter the source alias adding a prefix configured on each ISDN gw of all sites of routing of the call appropriately.

  Are there other ways cleaner to do? Remember that we do not want to change the habit of users, so they should not add the additional prefix.

  Thanks in advance for your answers.

  If you create a cluster, see it as a VCS. You might hit limits if you try to use them to separate areas.

  Dedicated (groups of) vcs in each are the cleanest way.

  Not to mention that, Yes, have a cpl on the source, a local sub area or else that can

  separate endpoints could route your specific sets of endpoints to ISDN specific gws.

• Wi - Fi calls stopped working

  I don't know if it was the update to iOS 9.3 or 9.2, but somewhere in there my vocation as a wi - fi stopped working.  I tried the fixed standard, i.e. the power off, the hard reset, active / disable the wifi calling router reset, etc, and nothing has worked.  Any idea would be appreciated.  Thank you very much.

  Device: iPhone 6

  Carrier: AT & T 24.1

  iOS Version 9.3.2

  Is recommended that the article below. I would add a reboot forced recommendations:

  Hold down the Home and Sleep/Wake buttons at the same time for about 15 seconds, until the Apple logo appears. You won't lose anything.

  Go to settings > phone > Wi - Fi calling and make sure Wi - Fi, the call is on.

  Restart your iPhone.

  To connect to another Wi - Fi network. Not all Wi - Fi networks work with Wi - Fi calling.

  Turn off, Wi - Fi calling then on again.

  Go to settings > general > Reset and tap reset network settings.

  Phone with Wi - Fi calling - Apple Support

• E1200 Cisco/Linksys router no warranty and I bought a month ago

  Can someone give me an email address so I can explain my situation? Or a contact name? Bought last week 19/12/13 and called router using the connection, they said it was an old model (I guess that based on S/N) and had no guarantee. I can't control the rotation of the store stocks, one and it bought in good faith, expecting that a guarantee only begins when you buy it. I paid to have it installed and will have to pay again to uninstall/reinstall to return to store and to fight with them. Who pays for this?

  Yes, we told the whole story, and offered at the reception by e-mail. The lady was adamant that there was no guarantee because it was an 'old' unit that had been in the warehouse of deposits House too long. Thanks for the email address, ill get my message just out of them.

• How to set up a '0' prior to call for outside calls (CCM 6.2.1)

  Community salvation.

  I think that this issue could be very easy to be decided by the experts of the calls voice cisco. What are the steps required to configure all the internal components of the phones that they must dial '0' to get an outside/international appeal of prior work? Is it a matter of templates for itinerary?

  Thanks in advance.

  Kind regards.

  Hello

  Yes, it is a RP "template routing." I hope to share with us your topology what your CUCM & gateway type (MGCP & H323). In any case regarding steps e, please find the below: -.

  1-configuration of your gateway that will be responsible for tree calls.on incoming and outgoing CUCM gateway device - select type.

  RG 2-group to configure or channel that will include your gateway recorded. CUCM call routing-road/hunting-lines group and assign your gateway listed

  3 - RL set which includes your RG that configured in step 2. Routing-road/hunting-other calls list, then assign the RG.

  3-model RP to configure or channel. for example

  0.161xxxxxxx for mobile

  0.00!                 for international

  At the partiton award

  B - RL that configured in step 3

  C - on the bottom, discard numbers "predot."

  Test; user dials a number that fits the RPS following 0.161xxxxxxx & 0.00!  the calls will go.

  Thank you

  Please note all relevant information

• RCK + RCKD in a context (ECC) extended call variable

  Hi all

  Any know if it is possible to get the key to call router (RCK) and router called key day (RCKD) in two separate variables ECC? How can it be done in the ICM scripting?

  Thank you & best regards

  Nick

  Here you go, good luck.

  David

• CVP newbie on routing questions

  Hello, newbie here in CVP. Got a deployment with UCCE/CVP 9 X and Conn CUCM/Unit 9 X. No Sip Proxy and the current context we have to route calls to CUCM prior routing to ICM/CVP. I know that's not the best Setup, but there is no budget to spend more managers to call for UnityC to CIM at this stage so all calls will be pretty closely from before CUCM hit ICM. That said, I'm getting conflicting information about how to configure routing. Probably more too read things. From my tests, I need 2 PIMS on every PG to cover both CVP servers which means I need 2 labels VRU to account for each customer routing of the CVP (ie. 121212 and 131313)? I have heard however I need one but my call routing will fail if it is not 2. Is this correct? If this is the case, how ICM knows what CVP VRU label to use when a call is initiated of CUCM to ICM? And how is that redundant PVC without proxy SIP or DNS Srv? Any recommendations would be greatly appreciated.

  No, you build the same label for all customers of routing CVP.

  Sent by Cisco Support technique iPhone App

• Change default handler calls for unity

  In 4.0.4 SR1 unit, how to change the default call handler, under the heading "Call Routing"? The default call manager section is grayed out and there is no way to change opening greeting, another manager of appeal.

  The predefined rules are gray because they are not intended to be changed. They are rules "rescue" which must be in place and valid at all times, otherwise it is possible that calls will not be managed properly.

  You can override default emergency rule basically call manager by simply adding a new rule above him, with the same criteria as the Default rule call Handler, but different ' send the call to the "destination. However, be sure tp change the order of the rule such that your new rule is just above the default rule call manager, but under the rule of Sign-In attempt. If it is above the rule of attempt to Sign-In, then the subscriber access simple message will not work. Whenever you make changes to the range tables you want to run tests to ensure that they wanted affect on different types of incoming calls.

  For more information on routing rules, see here:

  http://www.Cisco.com/en/us/products/SW/voicesw/ps2237/products_administration_guide_chapter09186a008022cb4e.html

• Port of native queue, the AA and the voice call

  Hi all

  Recently, I met a challenge problem, the customer got a busy tone when they called the number of hunting. We have AA 9 connection of the unit-based, we also have the queue of the native call (Hunt). When the client calls the DID number that we published, the number called will be translated to 100 on H.323 voice gateway, and then click CUCM and directly forward to voicemail (primarily to computer virtual in DN 100). Then the call will be forwarded to CUCM hunting through the connection of the unit to call manager after time control.

  Call routing as follows.

  Dial the number DID of PSTN--> voice gateway (translate PSTN number to 100)--> CUCM--> connection of the unit (call handler)--> native CUCM call queue

  We have a PRI dedicated to hunting, there are a large number of channels available when the question that's happened. In order to avoid congestion caused by the voice port. I added Subscriber connection of the unit so that the cluster CUC for 48-Port voice. Whenever I have reset the port in UC Group, the problem disappeared. Sometimes, there are only 3-5 calls for hunting, but the question is yet. I was wondering is there something wrong with the voice port settings?

  Thanks in advance,

  Eric

  I see, holiday would be difficult, but you can set the static days in the periods of time in CUCM, which is technically the same as in UCXN.

  I guess I see your description of what your question is exactly, are calls arriving is no longer the hunt group of?

• 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