Does someone know what SQL table is modified when an UCCE agent transfers a call using the finesse client, as well as where the call was transferred to? I can see the call in TCD, and I see the call disposition of "transfer" , but I do not see where it was transferred to. I am ultimately interested in creating a cuic report showing this
thanks
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I have cups8.x / CUCM8.6, migrating to imp/cucm10. Both presence servers are in the same domain. What do I need to read up on if I want to provide presence info between the two enviornments? Both enviornments will be up for some time while sites are individually migrated over.
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I have both sccp and sip phones where the Idivert key does not work. Basically doesnt do anything. only 1 active call, and with more than 1 active calls. Tried various flavors of 9.3.1Sr1 and Sr4 sccp and sip loads. Was just looking for traces so i could see if its hitting vm server.
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I have an analog line used with an fxo card. the analog line came with a local number on it. How do I config the voice port or dial peer on the router such that when using outbound calling , the recepient sees the originating number and not the analog number? ie IP phone 312-XXX-XXX , analog line is 847-XXX-XXXX. When I push a call out the line from my IP Phone, the recipient sees the call coming from 847, rather than my 312 number. thanks
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Hello. Im integrating Intrado E911 with Cube. I have two Intrado SBC's to route to. I need to route to the second one upon receiving a 486 user busy error. Would this be a sip profile? something like request 486 sip header to ?
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Hello George. I got it sorted. There were 2 diversion headers in the invite, the first coming from cucm, the 2nd from the sip profiles. The first was invalid, so its why it was failing. I unchecked "redirect diversion header-outbound" on the sip trunk, reset, and all is working. Correct diversion headers. Thank you for your quick reply and suggestions.
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If i dial the number, life is good, and the outside access code is stripped out. Would not the forwarded number also match the same dial peer, since its the same format?
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When forwarding a phone over a Cube to the pSTN, I get a 604 error. If I call the "forwarded to:" number directly, life is good. Is there something under voice service voip that I am missing?
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I will check the relevant switch info once I have access to it. I could tell from the show ephone unregistered that i was not getting a tftp address. I was asking to verify the process on the CME side, and that I followed the correct steps. Thank you for the debugs.
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I need to add two 7962 sccp phones to an existing CME. after i create the ephone, and ephone-dn, i have gone into telephony-service and entered the create-cnf files, but the phones still show as unregistered. Am I missing something? ephone-dn 81 octo-line number 581 label 581 description AJAY call-forward busy 4402599 call-forward noan 4402599 timeout 10 ephone 81 device-security-mode none mac-address 08CC.68E8.E3D4 ephone-template 1 type 7962 button 1:82
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Hello. Im looking for info on what the ERL and Jitter counters mean and how to relate them to any potential call quality problems. I found the following, but not sure what it should be ERL Echo return loss (in dB). ERL/Reflctr Time difference, in ms, between the original signal and the loudest echo (peak reflector) as detected by the echo canceller. Jitter Current values of the delay and the jitter of the packets around that delay.
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