06-19-2019 05:25 AM
Hello,
I've got a client who we recently installed a CME 12.0 system, 11 88xx phones, all SIP of course. They approached me with a question that I'm not too sure of the answer. When the receptionist uses Call Transferring, the alerting name (the caller display on the receiving phone) shows up as the reception's extension, and not the information of the call being transferred (Like the incoming call). I know that when you press the transfer button, it first attempts to establish a call between extension A and B, then clicking transfer a second time actually transfers the call.
My question is, is there a way to have the incoming number's information be sent instead of the extension details when transferring a call? Not sure if I'm wording this correctly or not...
06-19-2019 07:19 AM
To have the original-caller's callerID show up during transfer, configure "Blind Transfer". Here is how to do this for SIP CME:
CUCME Administration Guide - Configure Call Transfer on SIP Phones
Maren
06-19-2019 10:41 AM
I read that link you sent me. I guess there's not a simple way to just tell the system to always do a blind transfer?
06-19-2019 11:42 AM
For SCCP phones, there was a telephony-service command that would set the default as blind-transfer. For SIP phones, this must be accomplished via the template. The good news is that the templates can do a good deal more as well.
06-19-2019 08:44 AM - edited 06-19-2019 08:44 AM
Hi,
I think you are facing similar issue as mentioned in this post:
The solution on that post was to remove asserted-id ppi.
voice service voip
sip
no asserted-id ppi
Check that post for more details.
06-19-2019 09:46 AM
Good catch!
And do read the whole post to the end. Removing asserted-id ppi globally will affect other features such as call park and call pickup. In addition to removing it globally, you will want to add it at the dial-peer level for inbound external calls.
Maren
06-19-2019 10:43 AM
In doing that, it seems like some stuff like call parking is disabled.
Also, this system uses FXO ports for PSTN access, using connection plar opx (ext) as it's mean to put the call to the hunt group. Should I make an incoming dial-peer to solve this issue?
06-19-2019 11:46 AM
If you are not using SIP for PSTN access, I think you will need to leave the asserted-id ppi in place globally and go with templates for the blind transfer.
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