Routing of calls free of charge on CUCM on SIP trunk

Similar Questions

• Caller ID of VCS Jabber Video on SIP Trunk on the phone

  Hello, Netpros.

  I am trying to find a way to show a caller E.164 ID when a call to a video client Jabber a UCM or a PSTN phone.

  VCSC/E 7.1; 13.2 TMS, configured TMSPE.

  We have a VCSC > SIP Trunk > UCM (8,6) > H.323 GW w / PRI voice configured for the composition of and the PSTN destination.  We do this, so the Jabber client video can be used as a softphone, and so we can have transparent FindMe to voice and SNR to the video in either sense.  In TMS, we have a model of implementation configured with an address of the video, ILA and unit address.  He pulls ad office phone number for video of Jabber users.

  From what I read here:

  http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_FindMe_Express_Deployment_Guide_X7-1.pdf, we should be able to see the caller ID when you call send out an ISDN GW.

  «This section describes how to use FindMe with calls are routed through an ISDN gateway (for example, when you call a mobile phone, or any other destination accessible ISDN).» If VCS has Caller ID (Applications > FindMe > Configuration) defined to use the ID FindMe, the identification of the appellant presented will be the user E.164 phone number. »

  UCM SIP config is vanilla, by

  ttp: / /www.tandberg.com/collateral/documentation/Deployment_Guides/Cisco_VCS_Ci...

  However, when a video call Jabber is directed to a University Complutense of MADRID registered phone, we get "unknown caller".  When to call us directly to the PSTN, we see that the BTN from the PRI.

  What is strange, is that if we call another client of Jabber of VCS (with all the same problems), directly into a phone UCM via Internet on URI (for example [email protected] / * /) through the VCSE, we see the caller ID name.

  Thank you, all!

  Joshua

  Hi Joshua,.

  At present the VCS will only send caller E.164 id to a GW of H.323 ISDN, which is registered for VCS.

  There a feature request in the order book to define if E.164 Caller ID or name/Video address on a per-zone basis. This will also be sove the question when the call is to go on a SIP connection, as in this case, a remote ISDN GW.

  Thank you

  Guy

• CUCM v8.5 with 3 SIP Trunks to the Lync Server - Route algorithm of Distribution for the Group

  I CUCM connected to three different Lync server via 3 different SIP trunks.

  RG is composed of the following elements:

  LYNC SIP TRUNK 1 (1.1.1.1)

  LYNC SIP TRUNK 2 (2.2.2.2)

  LYNC SIP TRUNK 3 (3.3.3.3)

  The route group was built with "Top Down" the algorithm of distribution. The first SIP trunk knows congestion and some calls are never routed to secondary and tertiary SIP trunks.

  Based on all the forum posts I've seen - it seems that I have to configure the algorithm of group distribution of ranges as 'circular '.

  If I change the algorithm group of "Circular" lines - can I expect the following results:

  1. first call will go through LYNC SIP TRUNK 1

  2. second call will cross LYNC SIP TRUNK 2

  3. third call will cross LYNC SIP TRUNK 3

  When I change the algorithm of distribution of the route to the 'circular' group and click 'SAVE', I am invited on "RESET".  This service assigns to the existing calls through SIP Trunks?

  TIA,

  Amir

  Hello

  the circular algorithm will work the way you mentioned, but I suggest to try and make sure that is not perform integration

  to reset the SIP trunk actually active calls should not be made because the gose media directly between endpoint unles syou use something like trust rely wher evous enforce calls to go to the SIP

  Therefore, in most cases is should be fine just try configuration at the time of the calls will face service disruption

  hope this helps

• Video support for voice gateway router can call?

  Hi all

  My gateway voice (Cisco 2921-V/K9) router to connect to the PSTN using SIP Trunk provider. CUCM 11 enter my ipphone and videoconferencing (SX20).

  If someone who using video-conferencing (SX20) to call my (SX20) videoconferencing via dial my number RTC, it only not Visual voice call of appeal.

  I have question: If the router voice gateway support video call?, if it can how to configure and there is an additional module?

  Thank you

  Cyriac

  Hi Cyriac,

  For a video call from point to point, you can see the "Configure the video telephony" section of the following link

  http://www.Cisco.com/c/en/us/TD/docs/voice_ip_comm/CUCM/Admin/9_1_1/CCMS...

  If video conferencing Setup check the following "feature Deprecation announces for video Conferencing and transcoding using PVDM3 on ISR G2 product Bulletin"

  http://www.Cisco.com/c/en/us/products/collateral/unified-communications/...

  Manish

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• 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

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• Routing of calls based on the area Code

  I want to route the calls to some agents based on the caller's area code. I guess I'll need a database file to reference the calling party number. Then filter on the area code and the router accordingly. I never did look a script on a database. I would use a separate SQL database server for this.

  I can handle the routing part as soon as I know not if the number of matches. I just need some advice on getting to the database correctly.

  If I'm heading in the wrong direction? It must be an XML file instead? All the sample scripts or tips would be appreciated.

  No problem. Remember not all the nifty tools in the method Execute of Java later and I think that you will find very practical in many scenarios.

  Moreover, do not forget if you find a post is useful or solve your problem, you should give him a side :)

  Thank you

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  Hello

  Welcome to the Microsoft community!

  You can either reinstall the computer, which means it will have ANYTHING to this topic, or you can do a system restore when it was working fine.

  http://Windows.Microsoft.com/en-GB/Windows7/products/features/system-restore

  http://h10025.www1.HP.com/ewfrf/wc/document?cc=us&LC=en&docName=c01867418

  I hope that the above suggestions work!

  See you soon,.

  Leo

• 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.

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  Chat support: Mon - Fri 05:00-19:00 (US Pacific Time)<=== note="" days="" and="">

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