We have a UC540 and I have been trying to get CallerID blocking on a call-by-call basis.
From what I have reviewed online, the "caller-id block code *9" under "telephony-service" should provide my callers the ability to dial "*9" and then the outside line code "0" and then the number. This should block the CallerID for that call.
I have applied the configuration but the call fails when testing it.
I have tried apply SIP debug, but when debugging is activated by phone starts doing strange things when I try and make the test call, like when i pick up the handset it freezes. I suspect this is related to a memory issue, which is another issue. I digress.
We are using a SIP provider for PSTN access. According to them they support "anonymous callerID". Whether or not that means a caller ID of "null" is another question. Bottom line is I cant capture a SIP debug to diagnose where it is failing.
So then I started thinking, is there is a called-translation rule I could create to apply a "null" value to match case of a called "*9" etc.
Question 1: Has anyone had a similar issue regarding the block caller-id on a UC500 with SIP provider and founf a workaround?
Question 2: Can the caller-id issue be worked around by applying a translation rule to match the *9 but apply a null value? ..not sure how one would apply the null value in the translation rule?
Question 3: Is it possible to install more memory into a UC540?....cant really find a suitable SKU in this regard so I am not sure if a memory upgrade is supported?
I already have my dial-peers configured for my call legs. If I apply the "clid strip" command to my PSTN dial-peer then ALL my calls will not have a CallerID. This is not the requirement. I need only apply to apply the clid when the caller wants to hide the callerID.
Based on my research this should work.
Router# configure terminal
Router(config-telephony)# caller-id block code *9
Are you suggesting that I create another dial-peer for the same outgoing PSTN call leg? For example....
dial-peer voice 1234 voip
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling