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  • Users: chuck14
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  1. chuck14

    PRI (Qsig) Trunk to AudioCodes Mediant 3000 Calling Name not Being Sent to AudioCodes

    Hello I have a PRI (Qsig) trunk connecting to Mediant 3000 AudioCodes. When the opeartor receives the call and transfer it the orignating caller-id is not being sent on the transfered. The call still shows the operator name even after the transfer has been completed. In looking the sys logs in...
  2. chuck14

    Aura integrated with Salesforce

    Has anybody integrated the new Aura platform with Sales Forces? What I was thinking is with the One X appilcation is it possbile to have it integrated with the Salesforce web site? Therefore when anyone called the desktop if the anni infomation is collected it would pop the salesforce client...
  3. chuck14

    Aura Platform

    Hello we are looking at upgrading to the new Aura platform from CM3.1.4 I was hoping to get some feedback on the pro’s and con’s you may have experienced with the new architecture and the new features. Thanks in advance for your feedback
  4. chuck14

    has any one setup a skype gateway

    Is there a way to configure the skype gateway that would support any call from the pbx extension to any skype client over the internet and not use pstn. This would be the same for incoming calls. Incoming calls would be answer by a IVR then routed the correct ext. For example if we had a main...
  5. chuck14

    Genesys Call Center Needs to Route Calls to VM Direct

    Hello this may sound simple but having issues making this work correctly. We have a Genesys Call Center and I need to route calls to voicemail directly when the Analyst is unavailable. Which they could be on a call or has set themselves unavailable. Our environment is a little different from...
  6. chuck14

    Genesys Routing to Avaya VM Directly

    We are in the process in installing a Genesys solution. I am trying to get genesys to routing a call from an analyst directly to vm for the analyst. We have a Avaya CM 3.1 with module messaging. If genesys goes to an caller app they can't outpluse the right digits. I tried having it go to a...
  7. chuck14

    Polycom 6000 Soundstation

    Help, I have a 6000 Soundstation SIP phone register to the SES but I can't call outbound to any phone, inbound works. Any suggestions?????
  8. chuck14

    Exchange Integration w/96xx SIP

    Hello All, Is there away to have your exchange peresonal contacts integrated on the 96xx phones? I know you can do this using Softphone. Thanks
  9. chuck14

    DHCP Option 66

    Hello, I need help setting up Option 66 for the polycom soundstation IP 6000. I can't find anything that shows the fomrat it needs to enter the SIP server IP and other componets in DHCP. Thanks for your help, Chuck
  10. chuck14

    4690 Reg Error Wrong Set Type

    Hello, We have a Polycom 4690 that was working in another building. When we try to deploy it now it has a different ext. but the phone will not let us enter the new ext. The display shows "Reg Err Wrong Set Type" Is there away to change the ext on the phone? Thanks Chuck
  11. chuck14

    Polycom 6000 conference phones

    Hello, we are deploying about 30 polycom phones with our 96xx phones in a new building. We are using option 242 for the 96xx. Is there a dhcp option for the polycom 6000 sip phones??? Thanks
  12. chuck14

    Genesys SIP Routing to SES

    Hello, we are having issues with our Genesys SIP (GVP) routing to an external route point which is a vdn on the CM. I see the request hit the SES but it is not forwading the request to the CM. I have the media entry as ^:sip4753 point to the CM. Any help would be greatly appreciated. Chuck
  13. chuck14

    SES forwading to Genesys SIP server

    We are experiencing a busy signal when forwarding a call to ext. 4750 to SES that has a host entry to forward to the Genesys SIP. Not sure if the problem is on the SES or CM or Genesys. Thanks for your help
  14. chuck14

    ACD Question

    Is there away to have a agent go to aux work or after call when they finish a call. Currently all agents go to auto-in. Thanks
  15. chuck14

    4610 SIP Installation

    Hello, I am looking for a little help in configuring 4610 phone to work with our new SES. I have configured the SIP trunks but I am having issues trying to get the phone to work. If there is a link to provide instruction for setting up SIP phones, I would appreciate. Thanks
  16. chuck14

    Upgrade S8710 to CM 3.1.4

    Hello, I am doing a dot upgrade this weekend from CM 3.0.1 to 3.1.4 and I wanted to see if there are any techincal issues anyone has run into. It should be straight forward but I thought I would throw it out to everyone. Thanks for your help
  17. chuck14

    SIP Phones

    Is anybody using non Avaya SIP phones with 8710?????
  18. chuck14

    Speech Access

    We are looking at implementing the speech access server with our new module messaging platform. I understand the software is free but we need to build the server. The server requires a NMS Board per our business partner. Is there another option to the NMS board at a cheaper cost. The card is...
  19. chuck14

    LSP Mode

    I am configuring a S8300 w/G700 and I have it up and running. However when I disconnect the network link to the wan the S8300 will not go into LSP mode. What am I missing????? Thanks for your help.
  20. chuck14

    Installing a New S8300 LSP w/350

    Hello, I am configuring a G350 for the first time and the installation wizard is asking for the root password for the media gw. I tried root but it doesn't accept it. Does the 350 have a different root password? Thanks

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