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  1. chuck14

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

    here is what i got on private numb. This happens even if it is internal transfer. change private-numbering Page 1 of 1 NUMBERING - PRIVATE FORMAT Network Level: 0 PBX Identifier...
  2. chuck14

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

    it is already there and it happens on all transfer from any ext..
  3. chuck14

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

    thanks for the reply. The originating call-er is 909 213-37xx dial our main number AA dial 0 for the opeartor the opeartor transfer the call to my ext. 1578. My ext. is being forward to 641578. 641578 is configure in the UDP 64xxxx to route AAR 220. In running a trace on the pbx on tac 620...
  4. 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...
  5. 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...
  6. 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
  7. chuck14

    has any one setup a skype gateway

    thanks for the replies. this would not be using sip trunks but using the internet. busster can you provide the model numbers for the applicances. Thanks
  8. 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...
  9. chuck14

    Genesys Call Center Needs to Route Calls to VM Direct

    to add to the post above. i have a vdn point to a vector that has following command. 01 messaging skill 1 for extension active 02 stop I can get the call routed to vm but vm transfer it to the ext. instead of the mb for the ext. you enter. I have tweak the settign for extension field...
  10. chuck14

    IP Phone over ADSL

    we have several sites that have DSL for their broadband supporting IP phones. With our home office environment we have a cisco router providing the VPN tunnel. What we find was they need to have a minumum of 512K up and 1.5mb down speeds. Just remember it is the internet and you do not have...
  11. chuck14

    Avaya AES - switch status 'down'. Urgent, pls help

    do you have the switch side adminster? Has this every worked?
  12. 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...
  13. 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...
  14. chuck14

    Polycom 6000 Soundstation

    I found the problem. Even though the ext was register I had to assigned the ext. It's the only phone I had to set.
  15. 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?????
  16. 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
  17. 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
  18. chuck14

    4690 Reg Error Wrong Set Type

    thanks for the post. The PBX is the same it's the extension i need to change on the phone. It still has the old ext info but i can't get to the part to change the info. Chuck
  19. 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
  20. 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

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