CME 8.6 with IP phone and Analog Phone (Cisco 3825 + Vic2fxs )

malakipaa · ‎08-22-2014

Good day experts,

I am not an expert and would like to get help with my objectives.

What I have is IP phone and Analog phone connected to 3825 (CME 8.6) and connected to SIP ITSP.

What I want to do is for IP phone and Analog phone to ring at the same time and who ever answer first gets the call. The rests will stop ringing.

I have heard this is called Parallel Hunt in CME 8.6 but I would like verification for this.

I successfully setup to only redirect incoming call to FXs port but not to IP phone at the same time by using dialpeer. I hope this is possible since I have already invested to the router, fxs card and dsp card. I have a feeling that the easiest setup would be to get ATA box for analog phones.

Also can some one please send me a copy of your's config file for CME 8.6 that successfully configured that is similar to my objective? Like Dial peers, 911, and other stuff specifically for Toronto area if possible? I really appreciate if somebody can help me cause i have been pulling my hair lately. Thanks.

R0g22 · ‎08-23-2014

Hi,

Yes, you can configure a blast/parallel hunt group. This is what the configuration would look like for a hunt group -

n the following parallel hunt group example, when callers dial extension 1000, extension 1001, 1002, and so on ring
simultaneously. The first extension to answer is connected. If none of the extensions answers, the call is forwarded to
extension 2000, which is the number for the voice-mail service.

voice hunt-group 4 parallel
pilot 1000
list 1001, 1002, 1003, 1004
final 2000
timeout 20

You can tweak the above sample as per your requirements. Let me know if you got any questions related to this.

HTH,

Nipun

malakipaa · ‎08-23-2014

Hi, thanks I'll try this.

malakipaa · ‎08-23-2014

Hi Nipun, I have realized that the hunt group you have provided will only work for IP phone and not for analog phone via fxs port. Is there a way I can do this with cme 8.6? Thanks.

malakipaa · ‎08-23-2014

Hello all I have successfully converted analog phone into sccp. It has now its own extension number of 300. But now I need to be able to dial outside. How can I do this without assigning any of the phone with my full DID number with cme 8.6. I believe its called masking or translating my 3 digit extension into my 10 digit DID number so that my ITSP will allow me to route my call. Can somebody please help?

R0g22 · ‎08-27-2014

You are spot one.

You will need to create translation rule and profile.

Since you are trying to translate your 3 digit extensions to the 10 digit DID number, we can create a generic rule.

voice translation-rule 1

rule 1 /^...$/ /<the number that you need to translate to>/ (the forward slash is necessary)

voice translation-profile OUTBOUND_CALLER_ID

translate calling 1

Just apply the translation profile on an outbound dial-peer in the outgoing direction so that it translates your 3 digit extensions to full 10 digit DID numbers.

dial-peer voice x pots/voip

translation-profile outgoing OUTBOUND_CALLER_ID

You should be good at his point. Let me know if you need any clarifications.

HTH,

Nipun