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Testing outgoing calls with SIP provider and getting quite odd result.My topology:CUCM 6.1 - trunk - CUBE-- (FW) -- inet -- SIP provider -- PSTNWhen I call PSTN number via my SIP provider PSTN phone rings. When I answer the call is ended on PSTN end ...
I need to configure AA with couple different voice greetings depends of time, day. For example AA will have 2 greetings for business hours 9-17 and after, same will go for bank holidays or other free days. Best option would be calendar to use differe...
Getting error when I tried to make a call from SRST registered phone:Jul 14 17:50:58.954: ISDN Se0/3/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num 507316Jul 14 17:50:58.954: ISDN Se0/3/0:15 Q931: Sending SETUP callref = 0x008...
Got CUCM 6.1.3 with MGCP controlled gateway with access to PSTN.When I use softphone and g729 codec (marked option optimized for low bandwidth) all calls to PSTN are connected but both way no voice as g729 is a codec.Tried to use transcoder but still...
Can someone explain me a bit more about new 7.x feature called "dial via office".I found only: This feature allows users to make calls from their mobile phones as if they were dialing from their desk phones. Here: http://www.cisco.com/en/US/docs/voic...
You don't need External Phone Number Mask, second use traces to check if you hit this translation pattern. For testing I would use separate CSS and partition to make things clear.
Not sure what have you done so far but you can try with translation pattern: XXXX where you only change Calling Party Transform Mask to 0000 or whatever you want to use.
Yes, you can configure dial-peer for each voice port with translation.Something like this:number you will hide will be: 11111111 in this examplevoice translation-rule 1 rule 1 /.*/ /11111111/ !!voice translation-profile exit translate called 1!dial-p...
