Routing of calls

Similar Questions

• 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

  Jim

• Can not connect with the Adobe server to upgrade the PS. Tried with fiervall and the router and called my provider. She is also only CC there are problems!

  Can not connect with the Adobe server to upgrade the PS. Tried with fiervall and the router and called my provider. She is also only CC there are problems!

  Check your hosts file.

  help for that and other adobe.com sure common connection problems read, http://helpx.adobe.com/x-productkb/policy-pricing/activation-network-issues.html

• Photon: Car dock: routing of calls through car speakers turns off the phone mic?

  The title says it all, but I'll explain it. I bought the car dock and most of the time it looks cool, but there is a weird problem. I have the phone in the dock, and I have the audio cable from the charger to the auxiliary audio port of my car. When I try to make a call, the default speaker phone. In a noisy car environment, it is difficult to hear the other person of property. While the appeal goes, there is a "speaker" down button right on the phone that I can press on, which causes the audio to be routed via the speakers of the car being music, which is excellent. However, doing so causes both the person on the other line to be able to hear me, that makes no sense and is frankly very frustrating.

  I saw a couple of discussions on the issues, then I know that I'm not the only person to have had this experience, but I was not able to find a solution yet. Someone know if there is a solution, or if at least that there is a questions recognized by Motorola that could be fixed in the future? I guess it's a software issue any.

  Related discussions I've found:

  https://supportforums.Motorola.com/message/450352#450352 (reply 5)

  http://www.SprintUsers.com/Forum/archive/index.php/t-222513.html

  http://www.ramocafe.com/t318234.html

  Thank you

  Travis

  
  

• CTI Route Point Call Forwarding

  Hi all

  I'm running 10.5 CM and wanted to know if it was possible before/road called of a Point of ITC road to another CTI Route Point AND at the same time fighter pilot.

  We have just received a new switchboard console and it does not support the Points of CTI road instead, use groups of hunting/line to ring the DNs directly. We wanted to keep our old switchboard console along side again until we're done installation/check the other.

  One of my colleagues mentioned put the driver of hunting and the CTI Route Point in a hunt group and then reference the CTI route Point to that. Would this work?

  Please do not hesitate to provide alternatives that you think will work.

  Thank you

  Matt

  Hmm - you can't have a fighter pilot in the Group of the line.

  There are two options I can think.

  (1) first of all is to have your old extension of CTI - RP and new extensions of the monitoring console in a group of curves. You can then select the algorithm broadcast under fighter pilot group, if you want to send the call for both consoles.

  So you will have a new CTI - RP with call your everything to a fighter pilot. This fighter pilot will be a group of line containing all the extensions for the old and the new console monitoring

  But the downside will be to ring all extensions in the line fighter pilot group at the same time - which means both the old SIC - RP and all extensions for the new console at the same time.

  (2) the second option is to use number SNU. You can add the driver of hunting like the remote destination for the main extension. This will send the call to the old switchboard console and the fighter pilot at the same time - fighter pilot will then hunt for extensions according to the configuration (it will not be broadcast, can be circular, top-down etc. According to your needs). Drawback here is THAT SNR also feeds a license essential or fundamental.

  Others will also be consistent with more options I hope.

  -Terry

  Please note all useful posts

• 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

• Routing of calls free of charge on CUCM on SIP trunk

  Can @Mohammed al Baqari bro you help me with this one?

  There are a few possibilities here...

  Most toll-free numbers play a kind of announcement until the call is connected. It is possible that your provider should PRACK lying and that you also enable PRACK on CUCM. To do this, your provider must send 183 with SDP and request indicate BOF PRACK

  So contact your provider and explain the situation to them and know what is happening.

• Based on the sources, the routing of calls on VCS - E

  Hello

  We have now several customers, who can dial only an IP address. Is there a way to do a based ip source routing?

  Example:

  source ip address: 1.1.1.1. dest ip: 10.10.10.10 (VCS-E)--> translated destination: [email protected] / * /

  source ip address: 2.2.2.2. dest ip: 10.10.10.10(VCS-E)--> translated destination: [email protected] / * /

  source ip address: 3.3.3.3. dest ip: 10.10.10.10(VCS-E)--> translated destination: [email protected] / * /

  Translated destinations are known by the VCS-E.

  Best regards

  Daniel

  Maybe you're using the CPL, take a look at the start on pg 345 VCS Admin Guide .

  What endpoints they use and how they are deployed that restrict them to compose only via IP address?

• can anyone help? I was told the other day one can route your incoming calls by those iMac is a reality?

  You can route your calls through your iMac please.

  If you have an iPhone and the phone and iMac are on the same WiFi network, audio incoming calls ring also via Facetime. You can also make outgoing calls using Facetime audio, which are then sent through your phone.

  That's all assuming that your operating system software is up to date on both devices.

• Circumvention of toll - routing calls from China to voice GW United States

  Hi all

  I have an office in China and I'm configuring CUCM to route all U.S. calls China to voice gw office in the United States. Currently users in China face 9,001 and then registration number calls US. Users are used for numbering 9,001 for U.S. calls, so I like to keep the same numbering structure. How to configure CUCM for U.S. calls to voice gw routed to the United States in this scenario?

  I'm under CUCM 7.1.3 and voice gateways are H323

  Thanks to all in advance!

  D.

  No problem as long as the Gulf War send you the call of don't allows to route all calls as 11 digits. If the Gulf war is in place that requires composition 7 or 10 digits for calls local you must separate road of models for these area codes.

  HTH,

  Chris

• 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.

• How to call out are routed to a branch

  I'm looking for help in the understanding of outgoing calls how to a branch of work. I want to know what command/config on the router or call handler causes an outgoing call to be routed through a branch router fxo or trunk T1/SIP at Headquarters and what is shown on a recipients caller id. I have a new branch I want to when a call is made, it ask a FAC since the call manager at HQ but it is advisable to use the FXO lines and the recipient would only see the number associated with the line FXO and HQ not long-distance lines. What is happening now is sometimes the FXO lines are used and at other times, the line T1 to used headquarters. It appears however not sure algorithm that who and where this is done. See excerpts below. Any help would be appreciated.

  Circuit POTSGROUP group
  cause-code voice-class 1
  hunting-system round robin

  !
  !

  voice-port 1/0/0
  trunk-group POTSGROUP
  surveillance cut dualtone Mid-communication
  timeouts interdigit 7
  call waiting times - disconnect 3
  timeouts wait-version 3
  connection ERA opx 8032
  Description of the LINE of POTS
  activation of the caller ID
  !
  voice-port 0/0/2
  trunk-group POTSGROUP
  surveillance cut dualtone Mid-communication
  timeouts interdigit 7
  call waiting times - disconnect 3
  timeouts wait-version 3
  connection ERA opx 8032
  Description of the LINE of POTS
  activation of the caller ID
  !
  voice-port 0/0/3
  trunk-group POTSGROUP
  surveillance cut dualtone Mid-communication
  timeouts interdigit 7
  call waiting times - disconnect 3
  timeouts wait-version 3
  connection ERA opx 8032
  Description of the LINE of POTS
  activation of the caller ID
  !

  It gives me a better idea...

  Create levels and codes of the ACS.

  Create models of road and they point to the list of courses of your gateway or use SSS. On the model of routing it will tell you what college level to use.

  Your route list could contain the T1 to Headquarters under your FXO RG. This would allow the call to ride at T1 must be completed the FXO on this site.

  As long as the phones are registered with CUCM, they will have to use CAF. If this were the CME or SRST scenario this would not be the case.

  Here's a decent link on the configuration of the ACS:

  http://www.Cisco.com/c/en/us/support/docs/voice-unified-communications/u...

• Call the question of routing Manager

  Allows a calling Manager route to an extension based on the time of day?      For example, a call incoming during opening hours, calling press 0 and the call goes to the operator.    Incoming calls during the closing hours, calling press 0 and the call will the night watchman.      Is this possible?

  Thank you - Pat

  If you use Cisco Unity Connection, you can use the system lists to accomplish your question.  Basically, you say: 0 option to send a Manager to call "Operator called" or something else.  you attach a schedule to it and use the rules of transfer closed & STandard.  During the transfer of the normal working hours (hourly) to X and during the transfer closed at X.  Another way is to route the call to CUCM and using TOD routing in CUCM.  The easiest way is to use the unit.

• SPA-3102: how to fill the calls from SIP to PSTN and vice versa?

  Hello, since my ISP to my office blocked SIP ports, of I'll try and use it at home where the ATA works very well.

  If I call you ATA installed in my office in the United States (where there is no problem) to my ATA at home in India, I have how to route the call selectively to theFXS port or port FXO of the anti-terrorism Act in India? I I want to answer the call directly using the phone connected to the ATA instrument or make another local call out on the PSTN connection.

  TIA

  You get two accounts of Gizmo.  You put an account on the tab line 1, you put the other account on the PSTN line tab.  If you want to ring the phone attached to the SPA3102, you dial the account tab line 1.  If you want to get the tone to fill an outgoing line PSTN call, you call the account registered on the PSTN line tab.

• Could get a new router block printing? We had a lot of problems with the computer in the last week and a half. Bought new router and now the printer does not print. :-(

  (1) we went to DirecTV. Then the same day, our internet computer is down. I asked our current provider, Suddenlink, to let them know we had them to another - but we wanted to keep for the internet.

  (2) called DirecTV to investigate why our internet went down at the same time they installed? They said that they did nothing to cause that. We waited 5 days to make out and also install the third TV that was promised. The second tech gave things a lookover and said, once again, that there is nothing unusual during their passage to cause the problem.

  (3) called Suddenlink. They said it was the router. Called the people of Netgear. They said that the router was too old. We needed to pay an exorbitant sum for support and when I told him I'd rather just buy a new router and get free technical support for 90 days... and I personally do not have the credit cards in that time. (it's my mother s equip and I wonder it for the computer) when the guy told me that he could help and hung up on us, even though I was pleading for him to wait until I put my mother on the phone. (I'm in my 40s... not a kid).

  (4) Suddenlink called a few days later, when we got a new router (netgear brand NO!) Installed and I noticed that the 'ready' light was not 'on' on the modem. Their technology has said that he would check and call us in a few minutes. He did not.

  (5) the next day I called Suddenlink AGAIN. This time, I got a really nice customer service agent. She said she would send someone to check on it. It turns out that they had disconnected when I had called to pass only the TV.

  (6) in any case, we are finally online... only now... I can't the printer to work. :-(  This has worked before.  Could it be caused by the different router? I don't know about these things. I didn't even know what it was until someone at Radio Shack has explained to me. (I was not about to buy one at Wal-Mart - these poor people). The printer is a Canon Pixma MX340. We love him. The only thing is that it is difficult to clear the tasks or to reset. The only way to really do that is to simply pull the plug, unfortunately and restart the computer. But hey, it's a great printer, better than HP.

  We are under vista. The router is a Linksys wifi (which we have) N300 model E900.

  Of course I appreciate any help you can give. God bless you!

  Michelle and Ruth.

  Hello

  Thanks for posting in the Microsoft Community.

  I understand that you are facing problems with the printer in Windows Vista.

  I imagine the inconvenience that you are experiencing. I will definitely help you with this.

  Please follow the steps in the link.

  Solve printer problems

  http://Windows.Microsoft.com/en-in/Windows/printer-problems-in-Windows-help#fix-printer-problems=Windows-Vista&V1H=win8tab1&V2H=win7tab1&V3H=winvistatab1&v4h=winxptab1